From thangappan143 at gmail.com  Sat Aug  1 00:18:12 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 1 Aug 2009 12:48:12 +0530
Subject: [Freeswitch-users] ODBC problem
Message-ID: <7aa29e790908010018k71502d49ob69665a2d48b2047@mail.gmail.com>

While installing mod_lcr I got the following problem.
freeswitch at debian> load mod_lcr
API CALL [load(mod_lcr)] output:
-ERR [module load file routine returned an error]

freeswitch at debian> 2009-08-01 18:12:15 [INFO] mod_lcr.c:522
lcr_load_config() odbc_dsn is 192.168.1.222:freeswitch:freeswitch
2009-08-01 18:12:15 [INFO] mod_lcr.c:536 lcr_load_config() dsn is
"192.168.1.222", user is "freeswitch", and password is "freeswitch"
2009-08-01 18:12:15 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
found, and no default driver specified

2009-08-01 18:12:15 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
Database!
2009-08-01 18:12:15 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
config file
2009-08-01 18:12:15 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_lcr.so
**Module load routine returned an error**

What could the error?
How can I resolve it?

-- 
Regards,
Thangappan.M
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From dome at tel.co.th  Sat Aug  1 01:17:42 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 1 Aug 2009 15:17:42 +0700
Subject: [Freeswitch-users] ODBC problem
In-Reply-To: <7aa29e790908010018k71502d49ob69665a2d48b2047@mail.gmail.com>
References: <7aa29e790908010018k71502d49ob69665a2d48b2047@mail.gmail.com>
Message-ID: <8ccbff060908010117j6e2fe01ale4b72a12e70203a8@mail.gmail.com>

please check
/etc/odbc.ini
/opdbinst.ini

Dome C.

2009/8/1 Thangappan.M <thangappan143 at gmail.com>:
> While installing mod_lcr I got the following problem.
> freeswitch at debian> load mod_lcr
> API CALL [load(mod_lcr)] output:
> -ERR [module load file routine returned an error]
>
> freeswitch at debian> 2009-08-01 18:12:15 [INFO] mod_lcr.c:522
> lcr_load_config() odbc_dsn is 192.168.1.222:freeswitch:freeswitch
> 2009-08-01 18:12:15 [INFO] mod_lcr.c:536 lcr_load_config() dsn is
> "192.168.1.222", user is "freeswitch", and password is "freeswitch"
> 2009-08-01 18:12:15 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
> STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
> found, and no default driver specified
>
> 2009-08-01 18:12:15 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
> Database!
> 2009-08-01 18:12:15 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
> config file
> 2009-08-01 18:12:15 [CRIT] switch_loadable_module.c:839
> switch_loadable_module_load_file() Error Loading module
> /usr/local/freeswitch/mod/mod_lcr.so
> **Module load routine returned an error**
>
> What could the error?
> How can I resolve it?
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freeswitch at peely.com  Sat Aug  1 03:36:29 2009
From: freeswitch at peely.com (peely)
Date: Sat, 1 Aug 2009 03:36:29 -0700 (PDT)
Subject: [Freeswitch-users] LUA: Independent control of each call leg.
In-Reply-To: <87f2f3b90907311153n23a6a9e6h838206f281bc7d07@mail.gmail.com>
References: <24744087.post@talk.nabble.com>
	<87f2f3b90907311153n23a6a9e6h838206f281bc7d07@mail.gmail.com>
Message-ID: <24767998.post@talk.nabble.com>


Hi,

Thanks for your response. It's a real shame I can't get the async behavior I
want from Freeswitch/LUA as this is exactly the kind of abstraction layer I
hoped for in a SIP Application Server.

Most of the apps I want to develop would be served using the LUA environment
as-is but if a few scenarios I want to be able to perform a small amount of
activity whilst both legs are in a connected state, a prime example is a
b-leg "whisper" where you are still playing "ringing" to the a-party while
the b-party answers and hears a message just before connection to the
a-party.

Regards,


Neil.



mercutioviz wrote:
> 
> The level of control you need really isn't served by doing scripting from
> the dialplan. I highly recommend using ESL and the event socket. It will
> mean a bit of a paradigm shift in your coding, but with that shift comes a
> lot of power and control over what you can do with the calls - really
> limited only by your imagination.
> 
> -MC
> 
> On Fri, Jul 31, 2009 at 8:01 AM, peely <freeswitch at peely.com> wrote:
> 
>>
>> Hi,
>>
>> I'm trying to develop an application using lua and need to control the
>> inbound and outbound legs independently, even when they are switched
>> together.
>>
>> I can initiate the outbound session but I can't seem to bridge without
>> losing control of the script.
>>
>> Does anyone know a way I can allow ingress to egress calling whilst still
>> maintaining script control mid-call? I also need to ingress to hear
>> provisional speech during outbound connect. I've looked at conferencing
>> but
>> there seems to be quite a lot of automated messaging.
>>
>>
>>
>> Thanks,
>>
>>
>> Neil.
>>
> 
> 
> 

-- 
View this message in context: http://www.nabble.com/LUA%3A-Independent-control-of-each-call-leg.-tp24744087p24767998.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From krice at freeswitch.org  Sat Aug  1 04:52:16 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 01 Aug 2009 06:52:16 -0500
Subject: [Freeswitch-users] Connect to PostgreSQL database
In-Reply-To: <7aa29e790907312331v56770633m188a88aef5d4a9cc@mail.gmail.com>
Message-ID: <C69996A0.5A439%krice@freeswitch.org>

You need to look at using ODBC... That is what pretty much everything
uses... Connect FS to ODBC and ODBC to pgsql...



From: "Thangappan.M" <thangappan143 at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sat, 1 Aug 2009 12:01:36 +0530
To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Connect to PostgreSQL database

Dear all,

?I installed postgresql database in my machine. So now I need to connect the
database from freeswitch.When I searched about the site.They told , to load
the mod_lcr module.

I followed the following steps.
? * Edit the modules.conf file and uncommented the applications/mod_lcr line
?* make mod_lcr-install
?* edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and
uncomment the mod_lcr line.
?* Reload the freeswitch


Where I made a mistake?
Tell the steps to connect the postreSQL database.

-- 
Regards,
Thangappan.M


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From brian at freeswitch.org  Sat Aug  1 09:04:54 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 1 Aug 2009 11:04:54 -0500
Subject: [Freeswitch-users] LUA: Independent control of each call leg.
In-Reply-To: <24767998.post@talk.nabble.com>
References: <24744087.post@talk.nabble.com>
	<87f2f3b90907311153n23a6a9e6h838206f281bc7d07@mail.gmail.com>
	<24767998.post@talk.nabble.com>
Message-ID: <BCC8B127-B1A6-4AC6-B729-D9E389DBC0C9@freeswitch.org>

You can do all this via ESL-Lua cd libs/esl; make luamod

/b

On Aug 1, 2009, at 5:36 AM, peely wrote:

>
> Hi,
>
> Thanks for your response. It's a real shame I can't get the async  
> behavior I
> want from Freeswitch/LUA as this is exactly the kind of abstraction  
> layer I
> hoped for in a SIP Application Server.
>
> Most of the apps I want to develop would be served using the LUA  
> environment
> as-is but if a few scenarios I want to be able to perform a small  
> amount of
> activity whilst both legs are in a connected state, a prime example  
> is a
> b-leg "whisper" where you are still playing "ringing" to the a-party  
> while
> the b-party answers and hears a message just before connection to the
> a-party.
>
> Regards,
>
>
> Neil.



From gshfreesw at gmail.com  Sat Aug  1 09:35:09 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Sat, 1 Aug 2009 12:35:09 -0400
Subject: [Freeswitch-users] Connect to PostgreSQL database
In-Reply-To: <C69996A0.5A439%krice@freeswitch.org>
References: <7aa29e790907312331v56770633m188a88aef5d4a9cc@mail.gmail.com>
	<C69996A0.5A439%krice@freeswitch.org>
Message-ID: <5070fcbd0908010935n2d481397h210ce38057f03933@mail.gmail.com>

Thangappa,

Are you using Perl Event Socket? In that use, you can use Perl DBI to
connect to any DB instead of using the FS's ODBC module.

On Sat, Aug 1, 2009 at 7:52 AM, Ken Rice <krice at freeswitch.org> wrote:

>  You need to look at using ODBC... That is what pretty much everything
> uses... Connect FS to ODBC and ODBC to pgsql...
>
>
> ------------------------------
> *From: *"Thangappan.M" <thangappan143 at gmail.com>
> *Reply-To: *<freeswitch-users at lists.freeswitch.org>
> *Date: *Sat, 1 Aug 2009 12:01:36 +0530
> *To: *freeswitch-users <freeswitch-users at lists.freeswitch.org>
> *Subject: *[Freeswitch-users] Connect to PostgreSQL database
>
>
> Dear all,
>
>  I installed postgresql database in my machine. So now I need to connect
> the database from freeswitch.When I searched about the site.They told , to
> load the mod_lcr module.
>
> I followed the following steps.
>   * Edit the modules.conf file and uncommented the applications/mod_lcr
> line
>  * make mod_lcr-install
>  * edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and
> uncomment the mod_lcr line.
>  * Reload the freeswitch
>
>
> Where I made a mistake?
> Tell the steps to connect the postreSQL database.
>
> --
> Regards,
> Thangappan.M
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Sat Aug  1 09:43:28 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 01 Aug 2009 11:43:28 -0500
Subject: [Freeswitch-users] Connect to PostgreSQL database
In-Reply-To: <5070fcbd0908010935n2d481397h210ce38057f03933@mail.gmail.com>
Message-ID: <C699DAE0.5A45A%krice@freeswitch.org>

He specifically said from FreeSWITCH... The ONLY way to connect FreeSWITCH
directory to any database (this does not includes external scripts running
as a separate process using ESL) is to either a) write your own custom
module to do so or b) use ODBC on the modules that support that now



From: Shameem Shiek <gshfreesw at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sat, 1 Aug 2009 12:35:09 -0400
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Connect to PostgreSQL database

Thangappa,

Are you using Perl Event Socket? In that use, you can use Perl DBI to
connect to any DB instead of using the FS's ODBC module.

On Sat, Aug 1, 2009 at 7:52 AM, Ken Rice <krice at freeswitch.org> wrote:
> You need to look at using ODBC... That is what pretty much everything uses...
> Connect FS to ODBC and ODBC to pgsql...
> 
> 
> 
> From: "Thangappan.M" <thangappan143 at gmail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 1 Aug 2009 12:01:36 +0530
> To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Connect to PostgreSQL database
> 
> 
> Dear all,
> 
> ?I installed postgresql database in my machine. So now I need to connect the
> database from freeswitch.When I searched about the site.They told , to load
> the mod_lcr module.
> 
> I followed the following steps.
> ? * Edit the modules.conf file and uncommented the applications/mod_lcr line
> ?* make mod_lcr-install
> ?* edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and
> uncomment the mod_lcr line.
> ?* Reload the freeswitch
> 
> 
> Where I made a mistake?
> Tell the steps to connect the postreSQL database.
> 
> -- 
> Regards,
> Thangappan.M
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From darren at dmmhosting.co.uk  Sat Aug  1 03:53:06 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Sat, 1 Aug 2009 11:53:06 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>

I am considering using freeswitch and would like to know if this is possible.
 
The provider I use has a host that sits behind an OpenSER proxy. The hostname cannot get resolved by DNS on the internet.
 
Using freeswitch, at the moment, I am getting a DNS failure message for the host.
 
Is there a way of registering to this host and making calls through it by making all traffic go through the outbound proxy?
 
TIA
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4292 (20090730) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From gmaruzz at celliax.org  Sat Aug  1 11:34:34 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 1 Aug 2009 20:34:34 +0200
Subject: [Freeswitch-users] if using centos you should read this
In-Reply-To: <7b197bef0907302321p6aea0f2fhc542551b70392d6e@mail.gmail.com>
References: <6C1D282197D14BB9935770FD34B8F42A@noblesys.com> 
	<92D4AB13-73E0-4989-AC2F-A6703354D14B@freeswitch.org>
	<b16156850907302157p7898e599m8ae98c2b8f133d49@mail.gmail.com> 
	<87f2f3b90907302221u67ce4e32mf2e8faa8a4a13771@mail.gmail.com> 
	<b16156850907302236i696d3941i6f2a24ca59d9ff85@mail.gmail.com> 
	<7b197bef0907302321p6aea0f2fhc542551b70392d6e@mail.gmail.com>
Message-ID: <7b197bef0908011134u4ceeb820k13fdf6a222374235@mail.gmail.com>

http://linux.slashdot.org/story/09/08/01/1443221/CentOS-Administrator-Reappears

str8edge sends word that Lance Davis, the CentOS project administrator
who had mysteriously gone absent, has now returned and is working with
the development team to get things back on track. From their
announcement: "The CentOS Development team had a routine meeting today
with Lance Davis in attendance. During the meeting a majority of
issues were resolved immediately and a working agreement was reached
with deadlines for remaining unresolved issues. There should be no
impact to any CentOS users going forward. The CentOS project is now in
control of the CentOS.org and CentOS.info domains and owns all
trademarks, materials, and artwork in the CentOS distributions. We
look forward to working with Lance to quickly complete all the agreed
upon issues. More information will follow soon."

On Fri, Jul 31, 2009 at 8:21 AM, Giovanni Maruzzelli<gmaruzz at celliax.org> wrote:
> :-)!
>
>
> On Fri, Jul 31, 2009 at 7:36 AM, Muhammad
> Shahzad<shaheryarkh at googlemail.com> wrote:
>> Please read my email as,
>>
>>> CentOS has been a trusted platfrom for me from last 3+ years. I have
>>> developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
>>> boxes, and 27 out of 49 Asterisk box are still running on CentOS in
>>> production environment. I really wish and hope this great project continues.
>>>
>>> I don't know any of its developers personally but i am quite sure they
>>> will resolve their differences professionally and put this project back on
>>> track.
>>
>> This damn Google Spell made meaning of my entire post the possite. ;-(
>>
>> Thank you.
>>
>>
>> On Fri, Jul 31, 2009 at 11:21 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>>>
>>>
>>> On Thu, Jul 30, 2009 at 9:57 PM, Muhammad Shahzad
>>> <shaheryarkh at googlemail.com> wrote:
>>>>
>>>> CentOS has been a trusted platfrom for me from last 3+ years. I have
>>>> developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
>>>> boxes, and 27 out of 49 Asterisk box are still ruining on CentOS in
>>>> production environment. I really wish and hope this great project continues.
>>>>
>>>> I don't know any of its developers personally but i am quite sure they
>>>> will resolve their differences professionally and put this project back on
>>>> track.
>>>
>>> The guys doing the work have vowed to continue the project. The only real
>>> issues are who controls the centos.org domain name and how to handle
>>> donations to the project. CentOS isn't going anywhere but forward.
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From d at unwire.it  Sat Aug  1 11:57:35 2009
From: d at unwire.it (Darin Weeks)
Date: Sat, 1 Aug 2009 11:57:35 -0700
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>
References: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>
Message-ID: <989132e70908011157m67680efbm427282aaab87255a@mail.gmail.com>

Have you looked over various example configurations on the wiki?  see: SIP
Provider Examples

I'm a bit confused by your questions... to make voip calls work, you
basically need to register/authenticate with proxies/gateways to send and
receive calls through them.  Sounds like you need to find out from your
provider's documentation as to how they expect you to connect with them.
Why are you trying to connect to the host behind the proxy rather than the
proxy?


On Sat, Aug 1, 2009 at 3:53 AM, Darren Williams <darren at dmmhosting.co.uk>wrote:

>  I am considering using freeswitch and would like to know if this is
> possible.
>
>
>
> The provider I use has a host that sits behind an OpenSER proxy. The
> hostname cannot get resolved by DNS on the internet.
>
>
>
> Using freeswitch, at the moment, I am getting a DNS failure message for the
> host.
>
>
>
> Is there a way of registering to this host and making calls through it by
> making all traffic go through the outbound proxy?
>
>
>
> TIA
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4292 (20090730) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( (
broadband internet for west hollywood
   d at unwire.it        http://unwire.it
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From pjintheusa at gmail.com  Sat Aug  1 12:36:09 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 1 Aug 2009 15:36:09 -0400
Subject: [Freeswitch-users] LUA: Independent control of each call leg.
In-Reply-To: <24744087.post@talk.nabble.com>
References: <24744087.post@talk.nabble.com>
Message-ID: <367751820908011236p5db2fcfbu859bdbef60d6de7e@mail.gmail.com>

 >>a prime example is a b-leg "whisper" where you are still playing
"ringing" to the a-party while
>>the b-party answers and hears a message just before connection to the
>>a-party.

You should be able to do this particular function using group_confirm. See
this page in the wiki.

http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm

In your case the test.js would contain the whisper. Note that you can send
args to this script.


On Fri, Jul 31, 2009 at 11:01 AM, peely <freeswitch at peely.com> wrote:

>
> Hi,
>
> I'm trying to develop an application using lua and need to control the
> inbound and outbound legs independently, even when they are switched
> together.
>
> I can initiate the outbound session but I can't seem to bridge without
> losing control of the script.
>
> For example, if I use:
>
>
>        local api = freeswitch.API();
>        inSession = session;
>        inSession:answer();
>        inSession:setAutoHangup(false);
>
>
>        egSession = freeswitch.Session("sofia/default/mynum at mydomain.com");
>        egSession:setAutoHangup(false);
>
>        if egSession:ready() then
>                api:execute("uuid_bridge",inSession.uuid .. " " ..
> egSession.uuid);
>        end
>
>        while egSession:ready() do
>                inSession:sleep(1000);
>        end
>
> Then I lose the script entirely, and if I use:
>
>        inSession:execute("bridge", "sofia/default/mynum at mydomain.com")
>
> Then I lose the ability to control the call whilst the outbound is in
> progress.
>
> Does anyone know a way I can allow ingress to egress calling whilst still
> maintaining script control mid-call? I also need to ingress to hear
> provisional speech during outbound connect. I've looked at conferencing but
> there seems to be quite a lot of automated messaging.
>
>
>
> Thanks,
>
>
> Neil.
>
> --
> View this message in context:
> http://www.nabble.com/LUA%3A-Independent-control-of-each-call-leg.-tp24744087p24744087.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Aug  2 01:33:32 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Aug 2009 03:33:32 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <989132e70908011157m67680efbm427282aaab87255a@mail.gmail.com>
References: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>
	<989132e70908011157m67680efbm427282aaab87255a@mail.gmail.com>
Message-ID: <65978701-BC97-4C7F-9BC5-EC66F769F3ED@freeswitch.org>

Fill out proxy... with the fake hostname... then fill out register- 
proxy and/or outbound-proxy.

/b

On Aug 1, 2009, at 1:57 PM, Darin Weeks wrote:

> Have you looked over various example configurations on the wiki?   
> see: SIP Provider Examples
>
> I'm a bit confused by your questions... to make voip calls work, you  
> basically need to register/authenticate with proxies/gateways to  
> send and receive calls through them.  Sounds like you need to find  
> out from your provider's documentation as to how they expect you to  
> connect with them.  Why are you trying to connect to the host behind  
> the proxy rather than the proxy?
>
>
> On Sat, Aug 1, 2009 at 3:53 AM, Darren Williams <darren at dmmhosting.co.uk 
> > wrote:
> I am considering using freeswitch and would like to know if this is  
> possible.
>
>
> The provider I use has a host that sits behind an OpenSER proxy. The  
> hostname cannot get resolved by DNS on the internet.
>
>
> Using freeswitch, at the moment, I am getting a DNS failure message  
> for the host.
>
>
> Is there a way of registering to this host and making calls through  
> it by making all traffic go through the outbound proxy?
>
>
> TIA
>

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From markmorreny at gmail.com  Sun Aug  2 02:04:18 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sun, 2 Aug 2009 17:04:18 +0800
Subject: [Freeswitch-users] H248 support
Message-ID: <20ad6b920908020204p6a6bec6dt32f17638d778a4e0@mail.gmail.com>

Hi,

Someone told me to check here.  I am looking for a H248 supported gateway.
Does freeswitch support H248 or is there anyway to make it supportable in
freeSWITCH?

Thanks,
Mark
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From brian at freeswitch.org  Sun Aug  2 02:09:06 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Aug 2009 04:09:06 -0500
Subject: [Freeswitch-users] H248 support
In-Reply-To: <20ad6b920908020204p6a6bec6dt32f17638d778a4e0@mail.gmail.com>
References: <20ad6b920908020204p6a6bec6dt32f17638d778a4e0@mail.gmail.com>
Message-ID: <D489C23D-0858-4221-A37B-AB5AFF821B4C@freeswitch.org>

Not at this time.. you could try contacting consulting at freeswitch.org  
to see what it might take to gain this support by funding it.

/b

On Aug 2, 2009, at 4:04 AM, mark morreny wrote:

> Hi,
>
> Someone told me to check here.  I am looking for a H248 supported  
> gateway.  Does freeswitch support H248 or is there anyway to make it  
> supportable in freeSWITCH?
>
> Thanks,
> Mark



From darren at dmmhosting.co.uk  Sun Aug  2 03:31:17 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Sun, 2 Aug 2009 11:31:17 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <52d777d0-f93e-4e41-9293-d6c403f60080@dmmhosting.co.uk>

?and/or outbound proxy?, I do not seem to be able to find this parameter.
 
It is probably my terminology which is confusing too and what exactly I want to do. All I know is that on my Thomson ST2030. If I enter:
 
Registrar Server Address: bmnha-01.bt.com
Proxy Server Address: bmnha-01.bt.com
Outbound Proxy Server:www.bbvservice-560129.bt.com
 
Everything works fine, I would just like to be able to transfer this config to freeswitch.
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 02 August 2009 09:34
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
Fill out proxy... with the fake hostname... then fill out register-proxy and/or outbound-proxy.
 
/b
 
On Aug 1, 2009, at 1:57 PM, Darin Weeks wrote:



Have you looked over various example configurations on the wiki?  see: SIP Provider Examples

I'm a bit confused by your questions... to make voip calls work, you basically need to register/authenticate with proxies/gateways to send and receive calls through them.  Sounds like you need to find out from your provider's documentation as to how they expect you to connect with them.  Why are you trying to connect to the host behind the proxy rather than the proxy?


On Sat, Aug 1, 2009 at 3:53 AM, Darren Williams <darren at dmmhosting.co.uk> wrote:
I am considering using freeswitch and would like to know if this is possible.
 
The provider I use has a host that sits behind an OpenSER proxy. The hostname cannot get resolved by DNS on the internet.
 
Using freeswitch, at the moment, I am getting a DNS failure message for the host.
 
Is there a way of registering to this host and making calls through it by making all traffic go through the outbound proxy?
 
TIA
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4297 (20090801) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4297 (20090801) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From rdenert at tng.de  Sun Aug  2 05:18:50 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Sun, 2 Aug 2009 14:18:50 +0200 (CEST)
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <6CE276B1-75B1-4470-BDD0-0E51721D3658@jerris.com>
Message-ID: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>

Hello,

sorry but I don't understand your answer.

BR

----- Urspr?ngliche Mail -----
Von: "Michael Jerris" <mike at jerris.com>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] Language of the speech

we don;t have any german sound files?

On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:

> Hi again,
>
> does anybody know why my freeswitch doesn't play German speech-files?
>
> Here is my construct:
>
> I'm generating a random number in my lua script:
> rand = math.random(11, 1000);
>
> The I want that the freeswitch says the "random number":
> session:execute("say", "de name_spelled iterated " ..test_nummer);
>
> It works fine in English but I don't bring the freeswitch to say the  
> number in german. I installed the freeswitch-lang- 
> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
> <section name="phrases" description="Speech Phrase Management">
>    <macros>
>      <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>    </macros>
> </section>
>
> Here is an extraction from the fs_cli:
> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152  
> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid 
>  Execute set(default_language=de)
> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid 
>  SET [default_language]=[de]
>
> But I don't here any German speech only the English on
>
> Does anybody of you have an idea? Thanks a lot again.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
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Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
-
This e-mail may contain confidential and/or privileged information. If
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unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.


From mike at jerris.com  Sun Aug  2 11:06:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 2 Aug 2009 14:06:58 -0400
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
References: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
Message-ID: <A8ED0B8A-A1AD-4240-82AB-F5F125E2ACAA@jerris.com>

how is going to play files that we don't have?

On Aug 2, 2009, at 8:18 AM, Rudolf Denert wrote:

> Hello,
>
> sorry but I don't understand your answer.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Michael Jerris" <mike at jerris.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: Re: [Freeswitch-users] Language of the speech
>
> we don;t have any german sound files?
>
> On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
>
>> Hi again,
>>
>> does anybody know why my freeswitch doesn't play German speech-files?
>>
>> Here is my construct:
>>
>> I'm generating a random number in my lua script:
>> rand = math.random(11, 1000);
>>
>> The I want that the freeswitch says the "random number":
>> session:execute("say", "de name_spelled iterated " ..test_nummer);
>>
>> It works fine in English but I don't bring the freeswitch to say the
>> number in german. I installed the freeswitch-lang-
>> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
>> <section name="phrases" description="Speech Phrase Management">
>>   <macros>
>>     <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>   </macros>
>> </section>
>>
>> Here is an extraction from the fs_cli:
>> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152
>> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid
>> Execute set(default_language=de)
>> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid
>> SET [default_language]=[de]
>>
>> But I don't here any German speech only the English on
>>
>> Does anybody of you have an idea? Thanks a lot again.
>



From wiltingtree at gmail.com  Sun Aug  2 12:36:07 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sun, 2 Aug 2009 15:36:07 -0400
Subject: [Freeswitch-users] Bridging a call to an extension on another PBX.
Message-ID: <eb7797d70908021236x18eabbb4t6882553da99642a0@mail.gmail.com>

Hello,
I'm trying to conference-in a call from FreeSWITCH to an extension on
another PBX using sip.

According to the documentation, I think it should look like this:

       conference abc at default dial
{sip_auth_username=myuser,sip_auth_password=mypassword}sofia/external/
101 at 1.2.3.4

where 1.2.3.4 is the ip address of the remote pbx, and 101 is the extension.

I've tried adding a gateway for it in the sip profiles, and then doing this:

      conference abc at default dial sofia/mygateway/701

Both of these methods give me a result of:

      Call Requested: result: [DESTINATION_OUT_OF_ORDER]

I set-up my soft phone to register to the same ip address with the same
credentials, and it allows me to call the extension properly.

Can somebody please tell me what I'm doing wrong?

Thanks,
Adam
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From nik.middleton at noblesolutions.co.uk  Sun Aug  2 12:38:55 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Sun, 2 Aug 2009 20:38:55 +0100
Subject: [Freeswitch-users] Outbound socket question
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>

Hi Guys,

 

I'm using an outbound socket to control calls, and it works a charm.
However, what I'd like to do is send a custom event regarding the call
on hang-up.  The way I see things happening at the moment, and I could
be wrong, is that the socket is closed when a hang-up occurs, so am I
taking a chance trying to send the event then? (try to sneak out the
event before socket closure happens)  The other option is of course to
open an inbound socket and send the event, but I'd rather not do that if
possible.

 

Regards

  

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From jmesquita at gmail.com  Sun Aug  2 13:21:54 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 2 Aug 2009 17:21:54 -0300
Subject: [Freeswitch-users] Bridging a call to an extension on another
	PBX.
In-Reply-To: <eb7797d70908021236x18eabbb4t6882553da99642a0@mail.gmail.com>
References: <eb7797d70908021236x18eabbb4t6882553da99642a0@mail.gmail.com>
Message-ID: <5a8712120908021321n750e549cl3e2f994813cc3b2d@mail.gmail.com>

Adam,

Pastebin the logs. Also, a sip dump of both situations can really help.

To enable sip traces on FreeSWITCH all you have to do is type on the CLI:

sofia profile <profile> siptrace on/off


jmesquita

On Sun, Aug 2, 2009 at 4:36 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Hello,
> I'm trying to conference-in a call from FreeSWITCH to an extension on
> another PBX using sip.
>
> According to the documentation, I think it should look like this:
>
>        conference abc at default dial
> {sip_auth_username=myuser,sip_auth_password=mypassword}sofia/external/
> 101 at 1.2.3.4
>
> where 1.2.3.4 is the ip address of the remote pbx, and 101 is the
> extension.
>
> I've tried adding a gateway for it in the sip profiles, and then doing
> this:
>
>       conference abc at default dial sofia/mygateway/701
>
> Both of these methods give me a result of:
>
>       Call Requested: result: [DESTINATION_OUT_OF_ORDER]
>
> I set-up my soft phone to register to the same ip address with the same
> credentials, and it allows me to call the extension properly.
>
> Can somebody please tell me what I'm doing wrong?
>
> Thanks,
> Adam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From merul at mac.com  Sun Aug  2 14:34:08 2009
From: merul at mac.com (Merul Patel)
Date: Sun, 02 Aug 2009 22:34:08 +0100
Subject: [Freeswitch-users] Configuring Sangoma U100
Message-ID: <27E91460-3D50-4DF4-AF1D-95D97634112C@mac.com>

I'm new to FS, and experimenting with it on a constrained environment  
(PCEngines ALIX board running Voyage Linux 0.62).

So far, FS has compiled fine, and I can register multiple softphones  
and make calls between them, but I'm lost at how to configure a  
Sangoma U100 so I can make and receive calls over an analogue line.

I've installed the wanpipe drivers (3.5.4) from Sangoma, and the  
wanrouter utility detects the USB device, and I've compiled it to  
support the TDM API.

FS was compiled with the Openzap module - as best as I can tell.

I thought that I would be able to use the wancfg_tdmapi utility to  
configure the /etc/wanpipe/wanpipe1.conf and then use the generated  
configuration file as the basis for configuring autoload_configs/ 
openzap.conf.xml.

However, the wancfg utility doesn't generate the wanpipe1.conf, and  
I'm stumped.

Any pointers would be much appreciated.

Best regards,

Merul
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From msc at freeswitch.org  Sun Aug  2 15:35:01 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 2 Aug 2009 15:35:01 -0700
Subject: [Freeswitch-users] Outbound socket question
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
Message-ID: <87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>

On Sun, Aug 2, 2009 at 12:38 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  Hi Guys,
>
>
>
> I?m using an outbound socket to control calls, and it works a charm.
> However, what I?d like to do is send a custom event regarding the call on
> hang-up.  The way I see things happening at the moment, and I could be
> wrong, is that the socket is closed when a hang-up occurs, so am I taking a
> chance trying to send the event then? (try to sneak out the event before
> socket closure happens)  The other option is of course to open an inbound
> socket and send the event, but I?d rather not do that if possible.
>
Nik,

Perhaps the "linger" event socket command will do what you need? Check out
this commit:
http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009391.html

Let me know if it works for you and I'll be sure to get it documented
properly. If you get it working I'd love to see a code snippet so we can
wikify this knowledge. :)

Thanks,
MC
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From mike at jerris.com  Sun Aug  2 15:42:57 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 2 Aug 2009 18:42:57 -0400
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <27E91460-3D50-4DF4-AF1D-95D97634112C@mac.com>
References: <27E91460-3D50-4DF4-AF1D-95D97634112C@mac.com>
Message-ID: <4241B9E6-0F1F-4F9E-A1DF-CC0A1E7BE8F9@jerris.com>


On Aug 2, 2009, at 5:34 PM, Merul Patel wrote:

> I'm new to FS, and experimenting with it on a constrained  
> environment (PCEngines ALIX board running Voyage Linux 0.62).
>
> So far, FS has compiled fine, and I can register multiple softphones  
> and make calls between them, but I'm lost at how to configure a  
> Sangoma U100 so I can make and receive calls over an analogue line.
>
> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the  
> wanrouter utility detects the USB device, and I've compiled it to  
> support the TDM API.
>
> FS was compiled with the Openzap module - as best as I can tell.
>
> I thought that I would be able to use the wancfg_tdmapi utility to  
> configure the /etc/wanpipe/wanpipe1.conf and then use the generated  
> configuration file as the basis for configuring autoload_configs/ 
> openzap.conf.xml.

try wancfg_fs

> However, the wancfg utility doesn't generate the wanpipe1.conf, and  
> I'm stumped.
>
> Any pointers would be much appreciated.
>
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From brian at freeswitch.org  Sun Aug  2 16:10:48 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Aug 2009 18:10:48 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <52d777d0-f93e-4e41-9293-d6c403f60080@dmmhosting.co.uk>
References: <52d777d0-f93e-4e41-9293-d6c403f60080@dmmhosting.co.uk>
Message-ID: <1305B47F-256B-494A-B3D4-E1850E134A2C@freeswitch.org>

set the proxy, register-proxy to bmnha-01.bt.com and the outbound- 
proxy to www.bbvservice-560129.bt.com, I regard this type of config B  
R O K EN and the provider shouldn't be doing this with DNS names that  
do not exist in my opinion... if they are doing this for security sake  
they should just setup VPN or direct access... this type of setup must  
makes my skin crawl.

/b

On Aug 2, 2009, at 5:31 AM, Darren Williams wrote:

> ?and/or outbound proxy?, I do not seem to be able to find this  
> parameter.
>
> It is probably my terminology which is confusing too and what  
> exactly I want to do. All I know is that on my Thomson ST2030. If I  
> enter:
>
> Registrar Server Address: bmnha-01.bt.com
> Proxy Server Address: bmnha-01.bt.com
> Outbound Proxy Server:www.bbvservice-560129.bt.com
>
> Everything works fine, I would just like to be able to transfer this  
> config to freeswitch.

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From mrene_lists at avgs.ca  Sun Aug  2 18:11:11 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 2 Aug 2009 21:11:11 -0400
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
References: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
Message-ID: <B4C913FC-6332-489C-B0A1-887CBEECF96D@avgs.ca>

Da gibst keine eigentliches DE audio-Datei, nur text fuer text-to- 
speech, deshalb funktionniert es nicht.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 2-Aug-09 um 8:18 AM schrieb Rudolf Denert:

> Hello,
>
> sorry but I don't understand your answer.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Michael Jerris" <mike at jerris.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: Re: [Freeswitch-users] Language of the speech
>
> we don;t have any german sound files?
>
> On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
>
>> Hi again,
>>
>> does anybody know why my freeswitch doesn't play German speech-files?
>>
>> Here is my construct:
>>
>> I'm generating a random number in my lua script:
>> rand = math.random(11, 1000);
>>
>> The I want that the freeswitch says the "random number":
>> session:execute("say", "de name_spelled iterated " ..test_nummer);
>>
>> It works fine in English but I don't bring the freeswitch to say the
>> number in german. I installed the freeswitch-lang-
>> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
>> <section name="phrases" description="Speech Phrase Management">
>>   <macros>
>>     <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>   </macros>
>> </section>
>>
>> Here is an extraction from the fs_cli:
>> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152
>> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid
>> Execute set(default_language=de)
>> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid
>> SET [default_language]=[de]
>>
>> But I don't here any German speech only the English on
>>
>> Does anybody of you have an idea? Thanks a lot again.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -- 
> -- 
> Rudolf Denert, Technical Support
> TNG AG, NGN
> Projensdorfer Str. 324, D-24106 Kiel, Germany
> phone: +49 431 7097-10, fax: +49 431 7097-555
> mailto: rdenert at tng.de http://www.tng.de
> -
> Register: Amtsgericht Kiel, HRB 6596 KI
> Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
> Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
> Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
> Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
> -
> This e-mail may contain confidential and/or privileged information. If
> you are not the intended recipient (or have received this e-mail in
> error) please notify the sender immediately and destroy this e-mail.  
> Any
> unauthorized copying, disclosure or distribution of the material in  
> this
> e-mail is strictly forbidden.
>
> Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
> Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
> E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
> Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
> die unbefugte Weitergabe dieser Mail ist nicht gestattet.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From thangappan143 at gmail.com  Sun Aug  2 22:09:23 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 3 Aug 2009 10:39:23 +0530
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
Message-ID: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>

Dear all,

 I am not yet installed odbc in my machine for accessing the psql database.
I have got the following error while tried to load the mod_lcr command.

2009-08-03 15:45:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
found, and no default driver specified

2009-08-03 15:45:47 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
Database!
2009-08-03 15:45:47 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
config file
2009-08-03 15:45:47 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_lcr.so
**Module load routine returned an error**

I have set the correct driver informations in the odbc.ini(/etc and /home)
and odbcinst.ini(/etc).
I have create the symbolic links in the /usr/loca/freeswitch/etc/

Where I made a problem?

After some time I found that there is no spidermonket_odbc.so file.This has
been found while executing the freeswitch command. The error is,

2009-08-03 15:26:06 [ERR] mod_spidermonkey.c:931 sm_load_file() Error
Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
**/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open shared
object file: No such file or directory**

So please help me?
I am doing this for more than 8 hours.





-- 
Regards,
Thangappan.M
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From mattdfong at gmail.com  Sun Aug  2 22:22:47 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 2 Aug 2009 22:22:47 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
Message-ID: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>

A few users of mine have been getting hung-up on after leg b of the bridge
hangsups. I looked in the logs and they are being hungup with
an EXCHANGE_ROUTING_ERROR hangup cause. The problem is
only occurring intermittently and only when both leg a and leg b are both
passed thru external gateways (the problem does *not* exist if leg a is a
softphone connected directly to FS).
I'm wondering if there are any known incompatibilities with equipment out
there that would cause this error. Otherwise, I assume the best way to
diagnosis this further is to ngrep, which I'll do, but thought I'd ask here
first. Thanks.

--matt
hello hunter - hosted predictive dialer & voice broadcasting
http://www.hellohunter.com
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From mike at jerris.com  Sun Aug  2 23:12:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 01:12:51 -0500
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
In-Reply-To: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
References: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
Message-ID: <D1107F74-5E72-485C-89EF-ED66A3A1AFEC@jerris.com>

You need to first install unixodbc and it's related devel packages and  
then run configure and re build freeswitch.

On Aug 3, 2009, at 12:09 AM, "Thangappan.M" <thangappan143 at gmail.com>  
wrote:

> Dear all,
>
>  I am not yet installed odbc in my machine for accessing the psql  
> database. I have got the following error while tried to load the  
> mod_lcr command.
>
> 2009-08-03 15:45:47 [ERR] switch_odbc.c:164  
> switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC] 
> [Driver Manager]Data source name not found, and no default driver  
> specified
>
> 2009-08-03 15:45:47 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot  
> Open ODBC Database!
> 2009-08-03 15:45:47 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to  
> load lcr config file
> 2009-08-03 15:45:47 [CRIT] switch_loadable_module.c:839  
> switch_loadable_module_load_file() Error Loading module /usr/local/ 
> freeswitch/mod/mod_lcr.so
> **Module load routine returned an error**
>
> I have set the correct driver informations in the odbc.ini(/etc and / 
> home) and odbcinst.ini(/etc).
> I have create the symbolic links in the /usr/loca/freeswitch/etc/
>
> Where I made a problem?
>
> After some time I found that there is no spidermonket_odbc.so  
> file.This has been found while executing the freeswitch command. The  
> error is,
>
> 2009-08-03 15:26:06 [ERR] mod_spidermonkey.c:931 sm_load_file()  
> Error Loading module /usr/local/freeswitch/mod/ 
> mod_spidermonkey_odbc.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open  
> shared object file: No such file or directory**
>
> So please help me?
> I am doing this for more than 8 hours.
>
>
>
>
>
> -- 
> Regards,
> Thangappan.M
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mrene_lists at avgs.ca  Sun Aug  2 23:20:35 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 3 Aug 2009 02:20:35 -0400
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
Message-ID: <177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>

Hi,

Digging a bit in mod_sofia releaved that it can be caused by a SIP  
code 482 (loop detected), 483 (too many hops) or 484 (address  
incomplete).

Do a SIP trace to sched more light on what's happening.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong:

> EXCHANGE_ROUTING_ERROR



From thangappan143 at gmail.com  Mon Aug  3 00:07:29 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 3 Aug 2009 12:37:29 +0530
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
In-Reply-To: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
References: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
Message-ID: <7aa29e790908030007l1478b8b1mfa460a556349342f@mail.gmail.com>

I have installed unixodbc at first.But I am getting the same error.

On Mon, Aug 3, 2009 at 10:39 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>  I am not yet installed odbc in my machine for accessing the psql database.
> I have got the following error while tried to load the mod_lcr command.
>
> 2009-08-03 15:45:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
> STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
> found, and no default driver specified
>
> 2009-08-03 15:45:47 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
> Database!
> 2009-08-03 15:45:47 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
> config file
> 2009-08-03 15:45:47 [CRIT] switch_loadable_module.c:839
> switch_loadable_module_load_file() Error Loading module
> /usr/local/freeswitch/mod/mod_lcr.so
> **Module load routine returned an error**
>
> I have set the correct driver informations in the odbc.ini(/etc and /home)
> and odbcinst.ini(/etc).
> I have create the symbolic links in the /usr/loca/freeswitch/etc/
>
> Where I made a problem?
>
> After some time I found that there is no spidermonket_odbc.so file.This has
> been found while executing the freeswitch command. The error is,
>
> 2009-08-03 15:26:06 [ERR] mod_spidermonkey.c:931 sm_load_file() Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open shared
> object file: No such file or directory**
>
> So please help me?
> I am doing this for more than 8 hours.
>
>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From thangappan143 at gmail.com  Mon Aug  3 00:12:14 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 3 Aug 2009 12:42:14 +0530
Subject: [Freeswitch-users] Need Help In IVR
Message-ID: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>

Dear all,

I am in the process of implementing IVR in Perl using outbound socket.

In the case of the XML macro I can easily specify the timeout,inter digit
timeout value as
 <menu name="demo_ivr"
          greet-long="phrase:demo_ivr_main_menu"
          greet-short="phrase:demo_ivr_main_menu_short"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="4">
      <entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>
    </menu>

Is there any way for Perl to configure this values. Where are the variables
resides?
I am struggling to implement this?




-- 
Regards,
Thangappan.M
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From rdenert at tng.de  Mon Aug  3 00:57:37 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Mon, 3 Aug 2009 09:57:37 +0200 (CEST)
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <8128254.210221249286232840.JavaMail.root@zimbra.tng.de>
Message-ID: <14701697.210241249286257372.JavaMail.root@zimbra.tng.de>

Hallo,

ich habe also keine Chance mir eine Zahl vorlesen zu lassen, welche per Zufall generiere? :-/ Ich frage nur deshalb, da ich in einem Subordner diverse Soundfiles von Zahlen habe, welche ich von einem Debianpaket entpackt habe.

Gru? aus Kiel

Translation:

Hello,

there is no chance to read an random generated number from the freeswitch? I?m asking again because I have several german soundfiles of numbers which I extracted form a debianpacket. 

BR from Kiel

----- Urspr?ngliche Mail -----
Von: "Mathieu Rene" <mrene_lists at avgs.ca>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Montag, 3. August 2009 03:11:11 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] Language of the speech

Da gibst keine eigentliches DE audio-Datei, nur text fuer text-to- 
speech, deshalb funktionniert es nicht.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 2-Aug-09 um 8:18 AM schrieb Rudolf Denert:

> Hello,
>
> sorry but I don't understand your answer.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Michael Jerris" <mike at jerris.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: Re: [Freeswitch-users] Language of the speech
>
> we don;t have any german sound files?
>
> On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
>
>> Hi again,
>>
>> does anybody know why my freeswitch doesn't play German speech-files?
>>
>> Here is my construct:
>>
>> I'm generating a random number in my lua script:
>> rand = math.random(11, 1000);
>>
>> The I want that the freeswitch says the "random number":
>> session:execute("say", "de name_spelled iterated " ..test_nummer);
>>
>> It works fine in English but I don't bring the freeswitch to say the
>> number in german. I installed the freeswitch-lang-
>> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
>> <section name="phrases" description="Speech Phrase Management">
>>   <macros>
>>     <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>   </macros>
>> </section>
>>
>> Here is an extraction from the fs_cli:
>> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152
>> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid
>> Execute set(default_language=de)
>> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid
>> SET [default_language]=[de]
>>
>> But I don't here any German speech only the English on
>>
>> Does anybody of you have an idea? Thanks a lot again.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -- 
> -- 
> Rudolf Denert, Technical Support
> TNG AG, NGN
> Projensdorfer Str. 324, D-24106 Kiel, Germany
> phone: +49 431 7097-10, fax: +49 431 7097-555
> mailto: rdenert at tng.de http://www.tng.de
> -
> Register: Amtsgericht Kiel, HRB 6596 KI
> Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
> Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
> Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
> Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
> -
> This e-mail may contain confidential and/or privileged information. If
> you are not the intended recipient (or have received this e-mail in
> error) please notify the sender immediately and destroy this e-mail.  
> Any
> unauthorized copying, disclosure or distribution of the material in  
> this
> e-mail is strictly forbidden.
>
> Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
> Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
> E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
> Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
> die unbefugte Weitergabe dieser Mail ist nicht gestattet.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.


From merul at mac.com  Mon Aug  3 01:12:54 2009
From: merul at mac.com (Merul Patel)
Date: Mon, 03 Aug 2009 09:12:54 +0100
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <mailman.801.1249261906.2787.freeswitch-users@lists.freeswitch.org>
References: <mailman.801.1249261906.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <97532242-8290-47A9-900C-49E7567F4A4E@mac.com>

>> I'm new to FS, and experimenting with it on a constrained  
>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>
>> So far, FS has compiled fine, and I can register multiple  
>> softphones and make calls between them, but I'm lost at how to  
>> configure a Sangoma U100 so I can make and receive calls over an  
>> analogue line.
>>
>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the  
>> wanrouter utility detects the USB device, and I've compiled it to  
>> support the TDM API.
>>
>> FS was compiled with the Openzap module - as best as I can tell.
>>
>> I thought that I would be able to use the wancfg_tdmapi utility to  
>> configure the /etc/wanpipe/wanpipe1.conf and then use the generated  
>> configuration file as the basis for configuring autoload_configs/ 
>> openzap.conf.xml.
>
> try wancfg_fs

Thanks for the suggestion Michael, but the same result occurs as when  
I try wancfg_tdmapi, ie:

"No Sangoma voice compatible cards found/configured"

>
>> However, the wancfg utility doesn't generate the wanpipe1.conf, and  
>> I'm stumped.
>>
>> Any pointers would be much appreciated.
>>



From darren at dmmhosting.co.uk  Mon Aug  3 02:12:24 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 3 Aug 2009 10:12:24 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <2d4a177f-88f4-44e9-94d9-dae6235f718f@dmmhosting.co.uk>

Brian, this ?broken? business explains a lot, I just assumed this was a normal practise. This ?outbound-proxy? parameter, I don?t see any reference to this anywhere.
 
This just causes Registration Failed with status DNS Error [503]
 
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 03 August 2009 00:11
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
set the proxy, register-proxy to bmnha-01.bt.com and the outbound-proxy to www.bbvservice-560129.bt.com, I regard this type of config B R O K EN and the provider shouldn't be doing this with DNS names that do not exist in my opinion... if they are doing this for security sake they should just setup VPN or direct access... this type of setup must makes my skin crawl.
 
/b
 
On Aug 2, 2009, at 5:31 AM, Darren Williams wrote:



?and/or outbound proxy?, I do not seem to be able to find this parameter.
 
It is probably my terminology which is confusing too and what exactly I want to do. All I know is that on my Thomson ST2030. If I enter:
 
Registrar Server Address: bmnha-01.bt.com
Proxy Server Address: bmnha-01.bt.com
Outbound Proxy Server:www.bbvservice-560129.bt.com
 
Everything works fine, I would just like to be able to transfer this config to freeswitch.
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

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From raffaele.p.guidi at gmail.com  Mon Aug  3 04:43:01 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 3 Aug 2009 13:43:01 +0200
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
Message-ID: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>

I found the answer by myself while I had finished writing the e-mail. The
correct call url is loopback/<extension> (in this case the command
is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
future reference (can't find any example of that anywhere). Is the project
wiki accesible for anyone to contribute or do I have to ask for an
authorization?

Regards,
   Raffaele

*********** ORIGINAL QUESTION *************

Hi, I'm trying to call an extension wich is not associated to a user from
the ESL (or the CLI as well) using the "ORIGINATE" command.
Now, while

originate user/1001 1000


works perfectly

with:

originate user/fakecall 1000


I have an error:

2009-08-03 13:21:19.406250 [ERR] switch_ivr_originate.c:1494 Cannot create
outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]


...this is not surprising ("fakecall" of course is not an user), but I
cannot figure out what is the correct CALL URL for this extension. Same
error is reported using sofia/internal/fakecall. It seems that the
EXECUTE_EXTENSION method could do for the magic (it works when issued from
an other extension in the dialplan) but it is not available from the CLI nor
the event socket (I'm using a binary version for windows - freeswitch.msi -
dated jul, 11th 2009).

PS: Calling "fakecall" from a registered phone (or from portaudio) works as
expected.
PPS: the configuration of fakecall in dialplan/default.xml
...
<context name="default">
   <extension name="fakecall">
<condition field="destination_number" expression="^fakecall$">
 <action application="lua" data="lua/helloworld.lua test $1"/>
</condition>
 </extension>
....
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From chad at apartmentlines.com  Mon Aug  3 03:46:10 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Mon, 3 Aug 2009 06:46:10 -0400
Subject: [Freeswitch-users] arriving today for ClueCon
Message-ID: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>

i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this  
afternoon?  anybody need help with setup?


From juanbackson at gmail.com  Mon Aug  3 04:45:21 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Mon, 3 Aug 2009 19:45:21 +0800
Subject: [Freeswitch-users] Question about dynamic registration
Message-ID: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>

Hi,

Other than curl, is there anyway to do dynamic registration?
It there anyway to embed a script in freeswitch to do the authorization?

Thanks,
Anne
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From brian at freeswitch.org  Mon Aug  3 05:40:48 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 07:40:48 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <2d4a177f-88f4-44e9-94d9-dae6235f718f@dmmhosting.co.uk>
References: <2d4a177f-88f4-44e9-94d9-dae6235f718f@dmmhosting.co.uk>
Message-ID: <184C69C8-EBEA-487D-A921-B35C478F7607@freeswitch.org>

You must be on SVN trunk.

/b

On Aug 3, 2009, at 4:12 AM, Darren Williams wrote:

> Brian, this ?broken? business explains a lot, I just assumed this  
> was a normal practise. This ?outbound-proxy? parameter, I don?t see  
> any reference to this anywhere.
>
> This just causes Registration Failed with status DNS Error [503]
>

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From brian at freeswitch.org  Mon Aug  3 05:41:41 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 07:41:41 -0500
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
Message-ID: <6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>

Just look for large groups of people with laptops.  I'm sure you can't  
miss us.

/b

On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:

> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> afternoon?  anybody need help with setup?



From brian at freeswitch.org  Mon Aug  3 05:42:33 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 07:42:33 -0500
Subject: [Freeswitch-users] Question about dynamic registration
In-Reply-To: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
References: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
Message-ID: <9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>

You could build your own module to do it how ever you please.  But  
forking a script every time to auth is not very scalable.

/b

On Aug 3, 2009, at 6:45 AM, Juan Backson wrote:

> Hi,
>
> Other than curl, is there anyway to do dynamic registration?
> It there anyway to embed a script in freeswitch to do the  
> authorization?
>
> Thanks,
> Anne
> ______



From asannucci at gmail.com  Mon Aug  3 05:54:49 2009
From: asannucci at gmail.com (bakko)
Date: Mon, 3 Aug 2009 14:54:49 +0200
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
In-Reply-To: <7aa29e790908030007l1478b8b1mfa460a556349342f@mail.gmail.com>
References: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
	<7aa29e790908030007l1478b8b1mfa460a556349342f@mail.gmail.com>
Message-ID: <CC687DA31ED44F2B8541E3878F54CF7F@voztovoice>

If your linux distribution is Centos you have to install  unixODBC-devel postgresql-odbc unixODBC then compile freeswitch.

Look at this wiki page:

http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc

BR
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From msc at freeswitch.org  Mon Aug  3 05:59:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Aug 2009 05:59:34 -0700
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
Message-ID: <87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>

On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> I found the answer by myself while I had finished writing the e-mail. The
> correct call url is loopback/<extension> (in this case the command
> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
> future reference (can't find any example of that anywhere). Is the project
> wiki accesible for anyone to contribute or do I have to ask for an
> authorization?
>

All you need to do is sign up for a free account on the wiki and you can
start editing. It's a community resource and all FS users are invited to add
their respective knowledge.

As for not finding what you were looking for, does this page not have it?
http://wiki.freeswitch.org/wiki/Loopback

If not then please feel free to add to this page whatever your specific
scenario entails and give some examples.

-MC
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From a.afzali2003 at gmail.com  Mon Aug  3 06:38:03 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 3 Aug 2009 17:08:03 +0330
Subject: [Freeswitch-users] Missing mod_curl
Message-ID: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>

Hi,

I'll appreciate if somebody tell me where has gone the mod_curl ? I just
need to use it for http method calls.

Regards,
-- afshin
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From jgonzalez at sqli.com  Mon Aug  3 06:41:09 2009
From: jgonzalez at sqli.com (julien)
Date: Mon, 03 Aug 2009 15:41:09 +0200
Subject: [Freeswitch-users] Authentication problem when calling softphones
	from ipphones
Message-ID: <4A76E8F5.9070804@sqli.com>

Hello everyone,
I'm using a SIP trunk to link my PBX and FS.
My problem is when I try to call a softphone on FS from my ipphone, I've 
the following error on FS during Authentication :

2009-08-03 14:58:27.123817 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by 
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by 
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [WARNING] sofia_reg.c:1755 Can't find user 
[@[FS IP@]]
You must define a domain called '[FS IP@]' in your directory and add a 
user with the id="" attribute
and you must configure your device to use the proper domain in it's 
authentication credentials.

I defined my gateway to the PBX this way :

<include>
  <gateway name="pbxlyon">
  <param name="username" value="pbxlyon"/>
  <param name="realm" value="[PBX IP@]"/>
  <param name="password" value="pbxlyon"/>
  <param name="register" value="false"/>
  <param name="register-transport" value="udp"/>
  <param name="retry_seconds" value="300"/>
  </gateway>
</include>

I don't want the PBX to try to authenticate because I can't define a 
username nor a password for the authentication.
Thank for you time.
Best regards,
Julien GONZALEZ.







From mike at jerris.com  Mon Aug  3 06:59:35 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 08:59:35 -0500
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <97532242-8290-47A9-900C-49E7567F4A4E@mac.com>
References: <mailman.801.1249261906.2787.freeswitch-users@lists.freeswitch.org>
	<97532242-8290-47A9-900C-49E7567F4A4E@mac.com>
Message-ID: <FCA3464E-2A85-40F3-93AA-F8154470B399@jerris.com>

What is the output of wantouter hwprobe?

On Aug 3, 2009, at 3:12 AM, Merul Patel <merul at mac.com> wrote:

>>> I'm new to FS, and experimenting with it on a constrained
>>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>>
>>> So far, FS has compiled fine, and I can register multiple
>>> softphones and make calls between them, but I'm lost at how to
>>> configure a Sangoma U100 so I can make and receive calls over an
>>> analogue line.
>>>
>>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the
>>> wanrouter utility detects the USB device, and I've compiled it to
>>> support the TDM API.
>>>
>>> FS was compiled with the Openzap module - as best as I can tell.
>>>
>>> I thought that I would be able to use the wancfg_tdmapi utility to
>>> configure the /etc/wanpipe/wanpipe1.conf and then use the generated
>>> configuration file as the basis for configuring autoload_configs/
>>> openzap.conf.xml.
>>
>> try wancfg_fs
>
> Thanks for the suggestion Michael, but the same result occurs as when
> I try wancfg_tdmapi, ie:
>
> "No Sangoma voice compatible cards found/configured"
>
>>
>>> However, the wancfg utility doesn't generate the wanpipe1.conf, and
>>> I'm stumped.
>>>
>>> Any pointers would be much appreciated.
>>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mike at jerris.com  Mon Aug  3 07:03:08 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 09:03:08 -0500
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
Message-ID: <396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>

It is still there.

On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>  
wrote:

> Hi,
>
> I'll appreciate if somebody tell me where has gone the mod_curl ? I  
> just need to use it for http method calls.
>
> Regards,
> -- afshin
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From hoaianh at gmx.de  Mon Aug  3 07:30:44 2009
From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh)
Date: Mon, 03 Aug 2009 16:30:44 +0200
Subject: [Freeswitch-users] telnet to event socket
Message-ID: <4A76F494.5020101@gmx.de>

Hi,

I'm taking a close look at event socket on FS 1.0.3. Configuration is 
the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and 
fsconsole.pl work but I was not able to telnet to port 8021. As I've 
done that I received somewhat like:
#>auth/request

I typed in: auth ClueCon

After some seconds I've got the message 'connection close by foreign host'

Any ideas?

Thank you
Hoai-Anh


From vkozak at abisoft.spb.ru  Mon Aug  3 07:37:43 2009
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Mon, 3 Aug 2009 18:37:43 +0400
Subject: [Freeswitch-users] Fw: FreeSwitch doesn't play music on hold
	forbriged channel
Message-ID: <C6AE09778C1441F6842140EFFC55DF20@abisoft.biz>

No. I didn't. Moreover, I tried to set it explicitly using api uuid_setvar.
  ----- Original Message ----- 
  From: Kozak Vladimir 
  To: ?????? ?????? 
  Sent: Monday, August 03, 2009 5:28 PM
  Subject: Fw: [Freeswitch-users] FreeSwitch doesn't play music on hold forbriged channel



  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, July 31, 2009 5:22 PM
  Subject: Re: [Freeswitch-users] FreeSwitch doesn't play music on hold forbriged channel


  I don't see the variable hold_music ... did you remove it? 


  /b


  On Jul 31, 2009, at 5:24 AM, Kozak Vladimir wrote:



    The scenario is the following:
    FS User A dial an extension
    Extention opens outbound socket channel to my application
    My application bridges the call to FS User B
    The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
    The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
    User B is placed on hold but no music on hold is played to the caller (User A)


    I have outbound socket channel and the following sequence of commands/event:
    listening on [any] 8084 ...
    connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
    connect





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From rdenert at tng.de  Mon Aug  3 07:49:19 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Mon, 3 Aug 2009 16:49:19 +0200 (CEST)
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <4A76F494.5020101@gmx.de>
Message-ID: <2076715.215161249310959264.JavaMail.root@zimbra.tng.de>

Hi,

you only have to write "auth <password>" and hit enter twice. The default password is something like ClueCon.

BR

----- Urspr?ngliche Mail -----
Von: "Ngo-Vi Hoai-Anh" <hoaianh at gmx.de>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Montag, 3. August 2009 16:30:44 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: [Freeswitch-users] telnet to event socket

Hi,

I'm taking a close look at event socket on FS 1.0.3. Configuration is 
the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and 
fsconsole.pl work but I was not able to telnet to port 8021. As I've 
done that I received somewhat like:
#>auth/request

I typed in: auth ClueCon

After some seconds I've got the message 'connection close by foreign host'

Any ideas?

Thank you
Hoai-Anh

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
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From dujinfang at gmail.com  Mon Aug  3 07:51:06 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 3 Aug 2009 22:51:06 +0800
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <4A76F494.5020101@gmx.de>
References: <4A76F494.5020101@gmx.de>
Message-ID: <645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>


On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
> Hi,
>
> I'm taking a close look at event socket on FS 1.0.3. Configuration is
> the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
> fsconsole.pl work but I was not able to telnet to port 8021. As I've
> done that I received somewhat like:
> #>auth/request
>
> I typed in: auth ClueCon
>
followed by two Enters (\n\n).

> After some seconds I've got the message 'connection close by foreign  
> host'
>
> Any ideas?
>
> Thank you
> Hoai-Anh
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From patj at linklocal.net  Mon Aug  3 08:31:11 2009
From: patj at linklocal.net (Pat Jensen)
Date: Mon, 3 Aug 2009 08:31:11 -0700
Subject: [Freeswitch-users] Authentication problem when calling
 softphones	from ipphones
In-Reply-To: <4A76E8F5.9070804@sqli.com>
References: <4A76E8F5.9070804@sqli.com>
Message-ID: <0F432CFDE6E44442BB35719DF1034F6D5F011ACFB7@ws2008.linklocal.net>

Julien,

Place an ACL entry with the IP address for your PBX in the following file:

/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml

 <list name="domains" default="deny">
<node type="allow" cidr="123.12.12.1/32"/>   <!-- your pbx ip -->
</list>

This should allow unauthenticated invites from your PBX to hit FS.  Hope this helps.

Pat
________________________________________
From: julien [jgonzalez at sqli.com]
Sent: Monday, August 03, 2009 6:41 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Authentication problem when calling softphones      from ipphones

Hello everyone,
I'm using a SIP trunk to link my PBX and FS.
My problem is when I try to call a softphone on FS from my ipphone, I've
the following error on FS during Authentication :

2009-08-03 14:58:27.123817 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [WARNING] sofia_reg.c:1755 Can't find user
[@[FS IP@]]
You must define a domain called '[FS IP@]' in your directory and add a
user with the id="" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.

I defined my gateway to the PBX this way :

<include>
  <gateway name="pbxlyon">
  <param name="username" value="pbxlyon"/>
  <param name="realm" value="[PBX IP@]"/>
  <param name="password" value="pbxlyon"/>
  <param name="register" value="false"/>
  <param name="register-transport" value="udp"/>
  <param name="retry_seconds" value="300"/>
  </gateway>
</include>

I don't want the PBX to try to authenticate because I can't define a
username nor a password for the authentication.
Thank for you time.
Best regards,
Julien GONZALEZ.









From a.afzali2003 at gmail.com  Mon Aug  3 09:01:34 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 3 Aug 2009 20:31:34 +0430
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
	<396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
Message-ID: <debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>

I've gotten 1.0.4pre9 , but i can not see it :(
-- afshin


On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris <mike at jerris.com> wrote:

> It is still there.
>
> On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
>
> > Hi,
> >
> > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > just need to use it for http method calls.
> >
> > Regards,
> > -- afshin
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Mon Aug  3 09:12:17 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 4 Aug 2009 00:12:17 +0800
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
	<396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
	<debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>
Message-ID: <A948E259-A60C-484C-9113-E79A5F0E4C64@gmail.com>


On Aug 4, 2009, at 12:01 AM, afshin afzali wrote:
> I've gotten 1.0.4pre9 , but i can not see it :(
> -- afshin
>

In trunk.

>
> On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> It is still there.
>
> On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
>
> > Hi,
> >
> > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > just need to use it for http method calls.
> >
> > Regards,
> > -- afshin
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From a.afzali2003 at gmail.com  Mon Aug  3 09:19:17 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 3 Aug 2009 20:49:17 +0430
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <A948E259-A60C-484C-9113-E79A5F0E4C64@gmail.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
	<396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
	<debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>
	<A948E259-A60C-484C-9113-E79A5F0E4C64@gmail.com>
Message-ID: <debff5a80908030919k7f182f3aifef607846dc9fb88@mail.gmail.com>

Thanks

On Mon, Aug 3, 2009 at 8:42 PM, Seven Du <dujinfang at gmail.com> wrote:

>
> On Aug 4, 2009, at 12:01 AM, afshin afzali wrote:
> > I've gotten 1.0.4pre9 , but i can not see it :(
> > -- afshin
> >
>
> In trunk.
>
> >
> > On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris <mike at jerris.com>
> > wrote:
> > It is still there.
> >
> > On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>
> > wrote:
> >
> > > Hi,
> > >
> > > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > > just need to use it for http method calls.
> > >
> > > Regards,
> > > -- afshin
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From darren at dmmhosting.co.uk  Mon Aug  3 10:25:37 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 3 Aug 2009 18:25:37 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <718419f1-d02d-4623-bd5a-7852608df505@dmmhosting.co.uk>

Definitely the same reply:
 
FreeSWITCH Version 1.0.trunk (14457)
 
[ERR] sofia_reg.c:1460 05061292117 Registration Failed with status DNS Error [503]. failure #1 and so on.
 
The complete config is:
 
<include>
  <gateway name="05061292117">
  <param name="username" value="05061292117"/>
  <param name="realm" value="bmnha-01.bt.com"/>
  <param name="from-user" value="05061292117"/>
  <param name="from-domain" value="bmnha-01.bt.com"/>
  <param name="password" value="XXXXXXXXXXXXXXX"/>
  <param name="proxy" value="bmnha-01.bt.com"/>
  <param name="register-proxy" value="bmnha-01.bt.com"/>
  <param name="outbound-proxy" value="www.bbvservice-560129.bt.com"/>
  <param name="register" value="true"/>
  <param name="register-transport" value="udp"/>
  <param name="retry-seconds" value="10"/>
  </gateway>
</include>
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 03 August 2009 13:41
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
You must be on SVN trunk.
 
/b
 
On Aug 3, 2009, at 4:12 AM, Darren Williams wrote:



Brian, this ?broken? business explains a lot, I just assumed this was a normal practise. This ?outbound-proxy? parameter, I don?t see any reference to this anywhere.
 
This just causes Registration Failed with status DNS Error [503]
 
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From merul at mac.com  Mon Aug  3 10:56:43 2009
From: merul at mac.com (Merul Patel)
Date: Mon, 03 Aug 2009 18:56:43 +0100
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <mailman.816.1249310297.2787.freeswitch-users@lists.freeswitch.org>
References: <mailman.816.1249310297.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <EE4A46F0-42E4-4305-A2DC-EFC4F7401DE1@mac.com>

> What is the output of wantouter hwprobe?

voyage:~# wanrouter hwprobe

-------------------------------
| Wanpipe Hardware Probe Info |
-------------------------------
1 . U100       : BUSID=1-1 : V=00

Card Cnt: U100=1

voyage:~# dahdi_hardware
usb:001/002          wanpipe-     10c4:8461 Sangoma WANPIPE USB-FXO  
Device


>
> On Aug 3, 2009, at 3:12 AM, Merul Patel <merul at mac.com> wrote:
>
>>>> I'm new to FS, and experimenting with it on a constrained
>>>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>>>
>>>> So far, FS has compiled fine, and I can register multiple
>>>> softphones and make calls between them, but I'm lost at how to
>>>> configure a Sangoma U100 so I can make and receive calls over an
>>>> analogue line.
>>>>
>>>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the
>>>> wanrouter utility detects the USB device, and I've compiled it to
>>>> support the TDM API.
>>>>
>>>> FS was compiled with the Openzap module - as best as I can tell.
>>>>
>>>> I thought that I would be able to use the wancfg_tdmapi utility to
>>>> configure the /etc/wanpipe/wanpipe1.conf and then use the generated
>>>> configuration file as the basis for configuring autoload_configs/
>>>> openzap.conf.xml.
>>>
>>> try wancfg_fs
>>
>> Thanks for the suggestion Michael, but the same result occurs as when
>> I try wancfg_tdmapi, ie:
>>
>> "No Sangoma voice compatible cards found/configured"
>>
>>>
>>>> However, the wancfg utility doesn't generate the wanpipe1.conf, and
>>>> I'm stumped.
>>>>
>>>> Any pointers would be much appreciated.
>>>>



From dave at 3c.co.uk  Mon Aug  3 10:57:14 2009
From: dave at 3c.co.uk (David Knell)
Date: Mon, 3 Aug 2009 18:57:14 +0100
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
Message-ID: <01A26C17-A286-4A71-8B9C-8E558E963501@3c.co.uk>



-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



On 3 Aug 2009, at 13:41, Brian West <brian at freeswitch.org> wrote:

> Just look for large groups of people with laptops.  I'm sure you can't
> miss us.
>
> /b
>
> On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
>
>> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
>> afternoon?  anybody need help with setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Mon Aug  3 11:14:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 13:14:27 -0500
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <EE4A46F0-42E4-4305-A2DC-EFC4F7401DE1@mac.com>
References: <mailman.816.1249310297.2787.freeswitch-users@lists.freeswitch.org>
	<EE4A46F0-42E4-4305-A2DC-EFC4F7401DE1@mac.com>
Message-ID: <019E4A58-9EC6-4198-A041-DFE2C1219E14@jerris.com>

I have yet to configure one of these cards for freeswitch so it's  
possible it's not in the config util yet.  I suggest contacting  
sangoma support for assistance

On Aug 3, 2009, at 12:56 PM, Merul Patel <merul at mac.com> wrote:

>> What is the output of wantouter hwprobe?
>
> voyage:~# wanrouter hwprobe
>
> -------------------------------
> | Wanpipe Hardware Probe Info |
> -------------------------------
> 1 . U100       : BUSID=1-1 : V=00
>
> Card Cnt: U100=1
>
> voyage:~# dahdi_hardware
> usb:001/002          wanpipe-     10c4:8461 Sangoma WANPIPE USB-FXO
> Device
>
>
>>
>> On Aug 3, 2009, at 3:12 AM, Merul Patel <merul at mac.com> wrote:
>>
>>>>> I'm new to FS, and experimenting with it on a constrained
>>>>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>>>>
>>>>> So far, FS has compiled fine, and I can register multiple
>>>>> softphones and make calls between them, but I'm lost at how to
>>>>> configure a Sangoma U100 so I can make and receive calls over an
>>>>> analogue line.
>>>>>
>>>>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the
>>>>> wanrouter utility detects the USB device, and I've compiled it to
>>>>> support the TDM API.
>>>>>
>>>>> FS was compiled with the Openzap module - as best as I can tell.
>>>>>
>>>>> I thought that I would be able to use the wancfg_tdmapi utility to
>>>>> configure the /etc/wanpipe/wanpipe1.conf and then use the  
>>>>> generated
>>>>> configuration file as the basis for configuring autoload_configs/
>>>>> openzap.conf.xml.
>>>>
>>>> try wancfg_fs
>>>
>>> Thanks for the suggestion Michael, but the same result occurs as  
>>> when
>>> I try wancfg_tdmapi, ie:
>>>
>>> "No Sangoma voice compatible cards found/configured"
>>>
>>>>
>>>>> However, the wancfg utility doesn't generate the wanpipe1.conf,  
>>>>> and
>>>>> I'm stumped.
>>>>>
>>>>> Any pointers would be much appreciated.
>>>>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Mon Aug  3 11:13:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Aug 2009 11:13:51 -0700
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <01A26C17-A286-4A71-8B9C-8E558E963501@3c.co.uk>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
	<01A26C17-A286-4A71-8B9C-8E558E963501@3c.co.uk>
Message-ID: <87f2f3b90908031113u441f1091h63289b79f12ddc84@mail.gmail.com>

Excellent! Many of us are at the hotel already, getting everything set up.
-MC

On Mon, Aug 3, 2009 at 10:57 AM, David Knell <dave at 3c.co.uk> wrote:

>
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
>
> On 3 Aug 2009, at 13:41, Brian West <brian at freeswitch.org> wrote:
>
> > Just look for large groups of people with laptops.  I'm sure you can't
> > miss us.
> >
> > /b
> >
> > On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
> >
> >> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> >> afternoon?  anybody need help with setup?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug  3 11:16:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Aug 2009 11:16:33 -0700
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
References: <4A76F494.5020101@gmx.de>
	<645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
Message-ID: <87f2f3b90908031116i2ab7d7b5o38fda7ce8d9b36b5@mail.gmail.com>

On Mon, Aug 3, 2009 at 7:51 AM, Seven Du <dujinfang at gmail.com> wrote:

>
> On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
> > Hi,
> >
> > I'm taking a close look at event socket on FS 1.0.3. Configuration is
> > the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
> > fsconsole.pl work but I was not able to telnet to port 8021. As I've
> > done that I received somewhat like:
> > #>auth/request
> >
> > I typed in: auth ClueCon
> >
> followed by two Enters (\n\n).
>
> > After some seconds I've got the message 'connection close by foreign
> > host'
> >
>

Don't forget that you can change the password by modifying the value in
freeswitch/conf/autoload_configs/event_socket.conf.xml"

<param name="password" value="ClueCon"/>

-MC
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From testa at voicetechnology.com.br  Mon Aug  3 11:17:27 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 3 Aug 2009 15:17:27 -0300
Subject: [Freeswitch-users] FS beats Aculab Prosody S on subjective test
	on lay users for conference quality
In-Reply-To: <4A71AA86.90707@coppice.org>
References: <9cb0e15e0907290753q6ec3d45x680f5020a07699ed@mail.gmail.com>
	<1248886530.3818.2.camel@dk-d820> <4A70F593.5080204@coppice.org>
	<1248958990.4428.15.camel@dk-d820> <4A71AA86.90707@coppice.org>
Message-ID: <9cb0e15e0908031117p79355d51qf49821277d009f7d@mail.gmail.com>

Hi, a bit late on answering some of the questions, but, here we go.On
Aculab, all codecs were G711u. The same codec we have on FS:

freeswitch at conference> show channels
API CALL [show(channels)] output:
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure
c9cafeb8-803a-11de-8ceb-cb8648fd1ccf,inbound,2009-08-03
11:34:44,1249310084,sofia/internal/1000 at 192.168.0.40,CS_EXECUTE,Teste
Testa,1000,192.168.0.165,3200,conference,3200-192.168.0.40 at ultrawideband
,XML,default,L16,8000,PCMU,8000,
cc5edb90-803a-11de-8ceb-cb8648fd1ccf,inbound,2009-08-03
11:34:49,1249310089,sofia/internal/1000 at 192.168.0.40
,CS_EXECUTE,F.G.Testa,1000,192.168.0.249,3200,conference,3200-192.168.0.40 at ultrawideband
,XML,default,L16,8000,PCMU,8000,

2 total.

freeswitch at conference> conference list
API CALL [conference(list)] output:
Conference 3200-192.168.0.40 (2 members)
2;sofia/internal/1000 at 192.168.0.40
;cc5edb90-803a-11de-8ceb-cb8648fd1ccf;F.G.Testa;1000;hear|speak;0;0;300
1;sofia/internal/1000 at 192.168.0.40;c9cafeb8-803a-11de-8ceb-cb8648fd1ccf;Teste
Testa;1000;hear|speak|talking|floor;0;0;300

I think this answers some questions from Michael.
A packet dump I don't have right now.

Fernando G. Testa


On Thu, Jul 30, 2009 at 11:13 AM, Steve Underwood <steveu at coppice.org>wrote:

> David Knell wrote:
> > On Thu, 2009-07-30 at 09:21 +0800, Steve Underwood wrote:
> >
> >>
> >> High quality conferencing is a difficult task, and still a research
> >> topic. No two conferencing systems perform alike. The interesting thing
> >> about this and other reports is that the conferencing in Freeswitch is
> >> not very clever right now, yet people are already saying it beats
> >> various other offerings, including long time commercial offerings.
> >>
> >
> > It may well be that a simplistic implementation (noise gate, add them
> > all up) is all that's required for dealing with small groups or, more
> > generally, groups of any size which have a small number of active
> > speakers at any one time: it's predictable and unlikely to introduce
> > unpleasant side effects.
> >
> This is one of those situations where when you've experienced something
> better you make that your baseline for acceptability. I would consider a
> noise gate horribly crude, and VAD as the minimum for acceptable
> performance. If you've only used a noise gate you get used to it. If
> you're not sufficiently versed in the art you may well think nothing
> better is even possible.
>
> The fact that even the simple scheme, with noise gating, in Freeswitch
> is getting high praise, is pretty damning of mature commercial products.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From pgrondin at ip5.com  Mon Aug  3 11:53:11 2009
From: pgrondin at ip5.com (Patrick Grondin)
Date: Mon, 3 Aug 2009 14:53:11 -0400
Subject: [Freeswitch-users] Using tone_detect application
In-Reply-To: <6C525B211BA8F44CAC96A4F68333A9D119326E0813@VMBX107.ihostexchange.net>
References: <6C525B211BA8F44CAC96A4F68333A9D119326E0813@VMBX107.ihostexchange.net>
Message-ID: <6C525B211BA8F44CAC96A4F68333A9D119326E10A6@VMBX107.ihostexchange.net>

Hi,

What's the correct way to detect a number changed SIT tone in Freeswitch ?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Patrick Grondin
Sent: July-28-09 3:44 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Using tone_detect application

Hi,

I'm doing some tests between 2 FS to understand how the tone_detect application works.

I'm trying to detect a SIT tone, but I can't seem to detect the 3 tones. I only get the first activated tone.

If I have all tones activated - - - > I detect only the first tone of my wav file.
If I have 2nd segment tones and up activated - - - > I detect only the second tone of my wav file.
If I have the 3rd segment tones activated - - - > I detect only the third tone of my wav file.

I see that tone_detect can detect all 3 tones, but never at the same time. Does anyone have an idea of what I could be doing wrong ? Thanks !

I'm using FreeSWITCH Version 1.0.trunk (14397).

My dialplan looks like this :

    <extension name="incoming">
      <condition field="destination_number" expression="^(514[0-9][0-9][0-9][0-9][0-9][0-9][0-9])$">
        <action application="answer"/>
        <action application="tone_detect" data="SIT1LO 913.8 w +25000 set SIT1LO=true"/>   <!--1st segment low -->
        <action application="tone_detect" data="SIT1HI 985.2 w +25000 set SIT1HI=true"/>   <!--1st segment high -->
        <action application="tone_detect" data="SIT2LO 1370.6 w +25000 set SIT2LO=true"/>   <!--2nd segment low -->
        <action application="tone_detect" data="SIT2HI 1428.5 w +25000 set SIT2HI=true"/>   <!--2nd segment high -->
        <action application="tone_detect" data="SIT3LO 1776.7 w +25000 set SIT3LO=true"/>  <!--3rd segment low -->
        <action application="tone_detect" data="SITITU1 950 w +25000 set SITITU1=true"/>   <!--2nd segment low -->
        <action application="tone_detect" data="SITITU2 1400 w +25000 set SITITU2=true"/>   <!--2nd segment high -->
        <action application="tone_detect" data="SITITU3 1800 w +25000 set SITITU3=true"/>  <!--3rd segment low -->
        <action application="bridge" data="sofia/internal/$1 at 192.168.100.9"/>
      </condition>
    </extension>
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From raffaele.p.guidi at gmail.com  Mon Aug  3 12:13:14 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 3 Aug 2009 21:13:14 +0200
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
	<87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
Message-ID: <b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>

Yeah, it's there but, as you can see it from the google queries below it's
not easy to find that page - unless you know what exactly you are looking
for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
fill the gap (I'll be happy to do it ASAP)

http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=

http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=<http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=>
http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=

Also, this email now shows up first in googling for "non user extension
freeswitch" (which were the keywords I was looking for). I think this has
been useful! ;)

Many thanks and regards,
   Raffaele

On Mon, Aug 3, 2009 at 14:59, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> I found the answer by myself while I had finished writing the e-mail. The
>> correct call url is loopback/<extension> (in this case the command
>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>> future reference (can't find any example of that anywhere). Is the project
>> wiki accesible for anyone to contribute or do I have to ask for an
>> authorization?
>>
>
> All you need to do is sign up for a free account on the wiki and you can
> start editing. It's a community resource and all FS users are invited to add
> their respective knowledge.
>
> As for not finding what you were looking for, does this page not have it?
> http://wiki.freeswitch.org/wiki/Loopback
>
> If not then please feel free to add to this page whatever your specific
> scenario entails and give some examples.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Prometheus001 at gmx.net  Mon Aug  3 12:40:36 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 03 Aug 2009 21:40:36 +0200
Subject: [Freeswitch-users] TDM API: CMD: 18 : Operation not supported
Message-ID: <4A773D34.300@gmx.net>

Hello,

I setup libpri and a sangoma card A108DE, but I cannot dial out.
At startup I receive on the D channel
    TDM API: CMD: 18 : Operation not supported

When dialling Libpri debug shows that the numbering plan is fine and
that it accepts the screened number, but then it finally hangs up with:
       Ext: 1  Cause: Info. element nonexist or not implemented (99),
class = Protocol Error (e.g. unknown message) (6) ]

Does lipri sent any incorrect message here? Protocol is EuroISDN
(Q.931/Q.921).
Anybody has discovered this already?

I am on trunk 14419.

See debug and configs below.

Best regards
Peter

Starting FS:
2009-08-03 21:37:18.264829 [DEBUG] zap_io.c:2281 span 1 [d-channel]=[1:16]
TDM API: CMD: 18
: Operation not supported
2009-08-03 21:37:18.264915 [INFO] ozmod_wanpipe.c:287 configuring device
s1c16 as OpenZAP device 1:16 fd:55 DTMF: none
2009-08-03 21:37:18.264929 [DEBUG] zap_io.c:2281 span 1
[b-channel]=[1:17-31]

Dialling:
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Handling message
for SAPI/TEI=0/0
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- ACKing all
packets from 3 to (but not including) 4
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Since there was
nothing left, stopping T200 counter
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Stopping T203
counter since we got an ACK
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Nothing left,
starting T203 counter
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Protocol
Discriminator: Q.931 (8)  len=14
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Call Ref: len= 2
(reference 5/0x5) (Terminator)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Message type:
STATUS (125)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < [08 04 82 e3 98 74]
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Cause (len= 6) [
Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Public
network serving the local user (2)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 <                 
Ext: 1  Cause: Info. element nonexist or not implemented (99), class =
Protocol Error (e.g. unknown message) (6) ]
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 <             
Cause data 1: 98 (152, Non-Locking Shift To Codeset 0 IE)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 <             
Cause data 2: 74 (116, Redirecting Number IE)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < [14 01 01]
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Call State (len=
3) [ Ext: 0  Coding: CCITT (ITU) standard (0)  Call state: Call
Initiated (1)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Processing IE 8
(cs0, Cause)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Processing IE
20 (cs0, Call State)
2009-08-03 21:16:19.441416 [ERR] ozmod_libpri.c:88 Received unsolicited
status: Info. element nonexist or not implemented

openzap.conf
[span wanpipe PRI_1]
number => 1
trunk_type => E1
b-channel => 1:1-15
d-channel => 1:16
b-channel => 1:17-31

openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
   <settings>
     <param name="debug" value="10"/>
   </settings>
   <libpri_spans>
     <span name="PRI_1">
        <param name="node" value="cpe"/>
        <param name="dp" value="national"/>
        <param name="l1" value="alaw"/>
        <param name="switch" value="euroisdn"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="public"/>
      </span>
   </libpri_spans>




From brian at freeswitch.org  Mon Aug  3 14:24:11 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 16:24:11 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <718419f1-d02d-4623-bd5a-7852608df505@dmmhosting.co.uk>
References: <718419f1-d02d-4623-bd5a-7852608df505@dmmhosting.co.uk>
Message-ID: <C925BB2D-18A5-4741-95A3-F82D9B1F546C@freeswitch.org>

which of the hostnames is fake?

/b

On Aug 3, 2009, at 12:25 PM, Darren Williams wrote:

> Definitely the same reply:
>

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From darren at dmmhosting.co.uk  Mon Aug  3 14:39:00 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 3 Aug 2009 22:39:00 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <c857d204-9c47-4185-ba83-56737ac47462@dmmhosting.co.uk>

The bmnha-01.bt.com is the one that doesn?t resolve
www.bbvservice-560129.bt.com resolves to 62.239.15.140
 
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 03 August 2009 22:24
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
which of the hostnames is fake?
 
/b
 
On Aug 3, 2009, at 12:25 PM, Darren Williams wrote:



Definitely the same reply:
 
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From dule.maillist at gmail.com  Mon Aug  3 15:06:46 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Mon, 3 Aug 2009 18:06:46 -0400
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
	<87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
	<b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>
Message-ID: <914fc92a0908031506t285d2719need5f8937c0bb2a6@mail.gmail.com>

Technically, loopback is not meant to specifically allow you to call
non-user extensions, it simply allows you to hit the dialplan.
For example,

originate sofia/gw/gwname/fakecall 1000

(where gwname is the gateway routing your calls out, and 'fakecall' the
non-user extension)

will also allow you to dial to non-user extensions. This information would
be on any of the wiki pages detailing the originate command.

Dan


On Mon, Aug 3, 2009 at 3:13 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Yeah, it's there but, as you can see it from the google queries below it's
> not easy to find that page - unless you know what exactly you are looking
> for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
> fill the gap (I'll be happy to do it ASAP)
>
>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=
>
>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=
> <http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=
>
> Also, this email now shows up first in googling for "non user extension
> freeswitch" (which were the keywords I was looking for). I think this has
> been useful! ;)
>
> Many thanks and regards,
>    Raffaele
>
> On Mon, Aug 3, 2009 at 14:59, Michael Collins <msc at freeswitch.org> wrote:
>
>>
>>
>> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
>> raffaele.p.guidi at gmail.com> wrote:
>>
>>> I found the answer by myself while I had finished writing the e-mail. The
>>> correct call url is loopback/<extension> (in this case the command
>>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>>> future reference (can't find any example of that anywhere). Is the project
>>> wiki accesible for anyone to contribute or do I have to ask for an
>>> authorization?
>>>
>>
>> All you need to do is sign up for a free account on the wiki and you can
>> start editing. It's a community resource and all FS users are invited to add
>> their respective knowledge.
>>
>> As for not finding what you were looking for, does this page not have it?
>> http://wiki.freeswitch.org/wiki/Loopback
>>
>> If not then please feel free to add to this page whatever your specific
>> scenario entails and give some examples.
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From raffaele.p.guidi at gmail.com  Mon Aug  3 15:38:43 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Tue, 4 Aug 2009 00:38:43 +0200
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <914fc92a0908031506t285d2719need5f8937c0bb2a6@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
	<87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
	<b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>
	<914fc92a0908031506t285d2719need5f8937c0bb2a6@mail.gmail.com>
Message-ID: <b0a8792f0908031538g20abf2e0yba8dab8ac1283305@mail.gmail.com>

uhm... does it?

freeswitch at W2GZ8VNR01> originate sofia/gateway/callwithus.com/fakecall 1001
....
API CALL [originate(sofia/gateway/callwithus.com/fakecall 1001)] output:
-ERR CALL_REJECTED


Anyway, I clearly understand that loopback allows to "hit the dialplan",
maybe I couldn't find it before because of my poor english (but googling for
"freeswitch hit the dialplan" doesn't help either). Really, I simply had
problems finding it - the information was there (is it a matter of SEO -
Search Engine Optimization ;)?

Regards,
    Raffaele

On Tue, Aug 4, 2009 at 00:06, Dan Le <dule.maillist at gmail.com> wrote:

> Technically, loopback is not meant to specifically allow you to call
> non-user extensions, it simply allows you to hit the dialplan.
> For example,
>
> originate sofia/gw/gwname/fakecall 1000
>
> (where gwname is the gateway routing your calls out, and 'fakecall' the
> non-user extension)
>
> will also allow you to dial to non-user extensions. This information would
> be on any of the wiki pages detailing the originate command.
>
> Dan
>
>
> On Mon, Aug 3, 2009 at 3:13 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> Yeah, it's there but, as you can see it from the google queries below it's
>> not easy to find that page - unless you know what exactly you are looking
>> for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
>> fill the gap (I'll be happy to do it ASAP)
>>
>>
>> http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=
>>
>>
>> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=
>> <http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=>
>> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=
>>
>> Also, this email now shows up first in googling for "non user extension
>> freeswitch" (which were the keywords I was looking for). I think this has
>> been useful! ;)
>>
>> Many thanks and regards,
>>    Raffaele
>>
>> On Mon, Aug 3, 2009 at 14:59, Michael Collins <msc at freeswitch.org> wrote:
>>
>>>
>>>
>>> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
>>> raffaele.p.guidi at gmail.com> wrote:
>>>
>>>> I found the answer by myself while I had finished writing the e-mail.
>>>> The correct call url is loopback/<extension> (in this case the command
>>>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>>>> future reference (can't find any example of that anywhere). Is the project
>>>> wiki accesible for anyone to contribute or do I have to ask for an
>>>> authorization?
>>>>
>>>
>>> All you need to do is sign up for a free account on the wiki and you can
>>> start editing. It's a community resource and all FS users are invited to add
>>> their respective knowledge.
>>>
>>> As for not finding what you were looking for, does this page not have it?
>>> http://wiki.freeswitch.org/wiki/Loopback
>>>
>>> If not then please feel free to add to this page whatever your specific
>>> scenario entails and give some examples.
>>>
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Aug  3 20:36:19 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 22:36:19 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <c857d204-9c47-4185-ba83-56737ac47462@dmmhosting.co.uk>
References: <c857d204-9c47-4185-ba83-56737ac47462@dmmhosting.co.uk>
Message-ID: <A48DE4E2-997A-4E49-AB47-A12D5E9ABFF7@freeswitch.org>

Put the real hostname in register-proxy and outbound-proxy and the  
proxy needs to hold the fake one.

/b

On Aug 3, 2009, at 4:39 PM, Darren Williams wrote:

> The bmnha-01.bt.com is the one that doesn?t resolve
> www.bbvservice-560129.bt.com resolves to 62.239.15.140
>

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From thangappan143 at gmail.com  Mon Aug  3 22:47:03 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Tue, 4 Aug 2009 11:17:03 +0530
Subject: [Freeswitch-users] Fwd: Need Help In IVR
In-Reply-To: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>
References: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>
Message-ID: <7aa29e790908032247h2a93e838m1ac5fd3f8a3946c7@mail.gmail.com>

Can you please help me?


---------- Forwarded message ----------
From: Thangappan.M <thangappan143 at gmail.com>
Date: Mon, Aug 3, 2009 at 12:42 PM
Subject: Need Help In IVR
To: freeswitch-users <freeswitch-users at lists.freeswitch.org>


Dear all,

I am in the process of implementing IVR in Perl using outbound socket.

In the case of the XML macro I can easily specify the timeout,inter digit
timeout value as
 <menu name="demo_ivr"
          greet-long="phrase:demo_ivr_main_menu"
          greet-short="phrase:demo_ivr_main_menu_short"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="4">
      <entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>
    </menu>

Is there any way for Perl to configure this values. Where are the variables
resides?
I am struggling to implement this?




-- 
Regards,
Thangappan.M



-- 
Regards,
Thangappan.M
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From dujinfang at gmail.com  Tue Aug  4 00:17:32 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 4 Aug 2009 15:17:32 +0800
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story
Message-ID: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>

Hello All -   In the spirit of ClueCon (which we are missing this year, but
hopefully not next), we wanted to document our "FreeSWITCH Story".  We've
posted it to the wiki(
http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com) and
it is copied below.

Thank you all and enjoy a good conference!

Seven Du (seven)
Jonathan Palley (jpalley_idapted)
Idapted Ltd.


*How FreeSWITCH has created hundreds of job opportunities and changed
lives. *

We want to share our experience working with FreeSWITCH.  FreeSWITCH has
been a key enabler of our business.  We hope this story can be a small way
to say a very big THANK YOU ALL.

"Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
really allowed us to do just that.

What We Do:
We are not a telephony business; we are an educational technology and
service business. In Asia (China, in our case) students must pass English
examinations to study or work abroad and gain new experiences.  However,
there is limited access to native English speakers and the access students
can gain is typically very expensive.  At the same time, in the U.S., there
are many professionals looking for work-at-home opportunities - people who
need jobs and would create great teachers.  Through our technology and
content we empower these people to be effective English teachers.  Does it
work?  Yes.  The majority of our students are getting test scores that many
failed for years to get.  Just hours ago one student called one of our sales
agents crying with joy.  And for our teachers, they are now working in an
industry that was previously unavailable to those living in the U.S.
http://www.idapted.com

Why FreeSWITCH Enables This:
FreeSWITCH has been a key enabler of our business.  Recording calls,
controlling routing, integrating with various web-based interfaces, enabling
multiple endpoints - these are all key features of what we must do.  Most
importantly, setting up various servers and routes to mitigate cross-Pacific
and country-specific network challenges is key.  Doing what we are doing
with commercial solutions would have made the business unworkable.

Our Experiences with FreeSWITCH:
We started using FreeSWITCH as our VoIP Platform in April 2008, after
receiving unsatisfactory results with other open source solutions.  It took
one day of reading through the FreeSWITCH source code to know, "this is it.
 This is the VoIP platform we build our business on".  It took a few days of
working with the extremely competent and focused community to re-affirm this
commitment.

Our Setup:
Our teachers use a custom software that integrates a VoIP client with our
web based platform. Students connect to our teachers "on-demand".  Simply
put, on a web-based comet interface the student enters a phone number (or a
skype name or a gtalk account) and our platform bridges the best available
trainer and the student.  At the same time a web-based interface is being
updated.

The challenge for us is the connection between teachers and students over a
cross-continent network. For example, we experienced problems earlier this
year when a Asis-Pacific communication fiber broken... So, we've learned to
setup multi servers in multiple datacenters for redundancy.

We run multi instances of FreeSWITCH so we can always use the cutting edge
and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
only loads mod_dingaling. Here is one beauty of FS: We just had to create
different conf dirs (/usr/local/freeswitch, /usr/local/skype,
/usr/local/gtalk etc). This allows us to run the same code base over
different configurations, and call skype and gtalk accounts just like a
normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
abnormally or crashes, we can easily change to another FS-skype server (we
run other servers located in various places in China and HK for
redundancy).

FS --|
     |---PSTN gateways
     |--- FS-skype
     |--- FS-gtalk
     |--- FS-skype2
     |--- more ...



COMMUNITY:

The community's commitment cannot be undervalued.  The insightful, modular
design of FreeSWITCH allows anyone to contribute, whereever their skills
lie.  It also allows us to easily make modifications to the underlying code
to suit our specific use-cases  We want to highlight a few key people and
modules in the FS ecosystem:

mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
clients.  PSTN is zero-conf for the user and mitigates troubles with the end
users network/microphone, etc (which is significant with our user base).
 However, cheap providers fail randomly and FreeSWITCH's ability to control
routing, use multiple endpoints all while clearly seeing what is going on is
key.
Most importantly, anthm and the core team have been super helpful in getting
SIP to work with us.  Back in the pre 1.0 days anthm made significant
changes to mod-sofia to enable clients behind nats without STUN.  Its
important to point out that he didn't just make the changes -he forced us to
really make a compelling case as to why the changes were important for
FreeSWITCH.  This is a good thing.

skype (mod_skypiax): Due to the facts that users prefer skype, we configured
skypiax. It was unstable at the beginning and that's one of the reason we
started running that separate FS instance.  To be fair, it has caused a lot
of trouble - but we know this, its new software that takes a big risk and
implements a complex hack.  What is important is that the author of
skypiax(Giovanni Maruzzelli) has been a huge help. He's been very active
fixing bugs and logging in to our box to help trouble shoot. We owe him a
*big* thanks.

To make Skypiax more useful, we also created some patches including the ANY
and RR interfaces for sequential and round robin line hunting, some bug
fixes and other features like continue-load-on-fail and auto-skype-user
which haven't been merged into trunk yet. Thanks a community that gives us a
platform where we can all benefit and contribute.

erlang (mod_erlang_events): Another key enabler of the next release of our
system is the erlang interface.  We have a complex realtime queue routing
system has it handles input not just from freeswitch, but numerous other web
interfaces and sockets.  Erlang was the perfect technology to implement this
in and luckily an Erlang module for FreeSWITCH was already written.
Beautiful.

THE MORAL OF THE STORY:
FreeSWITCH is a great piece of software that has enabled new technologies
and business models.  The design has allowed (and the core team has
nurtured) a vibrant and exciting community that has made the software even
better.  Every day we go to work excited to push the boundaries of what can
be done with telephony technology and are confident this is the platform of
the future.

Thank you all.


Sincerely,

Du Jinfang (Seven) - Technical Operations/VoIP Manager
Jonathan Palley - CTO
Idapted Ltd.
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From hoaianh at gmx.de  Tue Aug  4 00:39:24 2009
From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh)
Date: Tue, 04 Aug 2009 09:39:24 +0200
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
References: <4A76F494.5020101@gmx.de>
	<645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
Message-ID: <4A77E5AC.9050501@gmx.de>

Hi,

Thank you. It works now. The clue is to hit 'Enter' twice.

Seven Du schrieb:
> On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
>   
>> Hi,
>>
>> I'm taking a close look at event socket on FS 1.0.3. Configuration is
>> the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
>> fsconsole.pl work but I was not able to telnet to port 8021. As I've
>> done that I received somewhat like:
>> #>auth/request
>>
>> I typed in: auth ClueCon
>>
>>     
> followed by two Enters (\n\n).
>
>   
>> After some seconds I've got the message 'connection close by foreign  
>> host'
>>
>> Any ideas?
>>
>> Thank you
>> Hoai-Anh
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From digitaldaz100 at gmail.com  Tue Aug  4 02:10:10 2009
From: digitaldaz100 at gmail.com (Darren Williams)
Date: Tue, 4 Aug 2009 10:10:10 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <26f14763-6f40-4468-91f3-e60a46892183@dmmhosting.co.uk>

This produces:
 
send 639 bytes to udp/[62.239.15.140]:5060 at 09:06:29.640488:
   ------------------------------------------------------------------------
   REGISTER sip:bmnha-01.bt.com SIP/2.0
   Via: SIP/2.0/UDP 91.121.159.57:5080;rport;branch=z9hG4bKQFFy1Ug5DcU0S
   Max-Forwards: 70
   From: <sip:05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
   To: <sip:05061292117 at bmnha-01.bt.com;transport=udp>
   Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
   CSeq: 118570058 REGISTER
   Contact: <sip:gw+05061292117 at 91.121.159.57:5080;transport=udp>
   Expires: 3600
   User-Agent: THOMSON ST2030 hw5 fw1.56 00-1F-9F-16-4E-99
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Content-Length: 0
 
   ------------------------------------------------------------------------
recv 726 bytes from udp/[62.239.15.140]:5060 at 09:06:29.656368:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 91.121.159.57:5080;received=91.121.159.57;branch=z9hG4bKQFFy1Ug5DcU0S;rport=5080
   From: <sip:05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
   To: <sip:05061292117 at bmnha-01.bt.com;transport=udp>;tag=SD567ec99-8a92eaf49d827b084b51e364ec68ae70.bc35
   Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
   CSeq: 118570058 REGISTER
   WWW-Authenticate: Digest realm="bmnha-01.bt.com", nonce="4a77fb4b6fe33ffccf77e6bdd6f7a7a397cdfd17", qop="auth"
   Server: Sip EXpress router (0.9.6 (sparc/solaris))
   Content-Length: 0
   Warning: 392 sip:5060 "Noisy feedback tells:  pid=9150 req_src_ip=172.20.92.61 req_src_port=5060 in_uri=sip:bmnha-01.bt.com out_uri=sip:bmnha-01.bt.com via_cnt==1"
 
   ------------------------------------------------------------------------
2009-08-04 11:06:29.655585 [ERR] sofia_reg.c:1460 05061292117 Registration Failed with status Operation has no matching challenge  [904]. failure #1 and so on.
 
Should my side not have responded at this point with another REGISTER with auth information in?
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 04 August 2009 04:36
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
Put the real hostname in register-proxy and outbound-proxy and the proxy needs to hold the fake one.
 
/b
 
On Aug 3, 2009, at 4:39 PM, Darren Williams wrote:



The bmnha-01.bt.com is the one that doesn?t resolve
www.bbvservice-560129.bt.com resolves to 62.239.15.140
 
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From markmorreny at gmail.com  Tue Aug  4 04:08:36 2009
From: markmorreny at gmail.com (mark morreny)
Date: Tue, 4 Aug 2009 19:08:36 +0800
Subject: [Freeswitch-users] event socket vs erlang
Message-ID: <20ad6b920908040408x48802b84mbf83e20a2cb5d2f@mail.gmail.com>

Hi,

I have seen people using both event socket and erlang to control freeSWITCH
externally.

What is the pros and cons of using event socket vs erlang?

Thanks,
Mark
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From rdenert at tng.de  Tue Aug  4 04:14:43 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Tue, 4 Aug 2009 13:14:43 +0200 (CEST)
Subject: [Freeswitch-users] Module in Lua not working
In-Reply-To: <19314518.221201249383746271.JavaMail.root@zimbra.tng.de>
Message-ID: <7203453.221221249384483704.JavaMail.root@zimbra.tng.de>

Hello again!

I need some help again, because I have little trouble with a few modules. The first one is luasocket, the seconde one is luasql. I always get the error:

error loading module 'socket' from file '/usr/local/lib/lua/5.1/socket/core.so': /usr/local/lib/lua/5.1/socket/core.so: undefined symbol: lua_getmetatable

and

error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_pushlstring

in the fs_cli

This looks like a conflict with an older version of these modules. Is this right? If yes, what should i update?

Thanks again.

BR

-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
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die unbefugte Weitergabe dieser Mail ist nicht gestattet.


From gmaruzz at celliax.org  Tue Aug  4 06:35:29 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 4 Aug 2009 15:35:29 +0200
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
Message-ID: <7b197bef0908040635k330c73cayc354748f40a3bf19@mail.gmail.com>

Cool!



On Tue, Aug 4, 2009 at 9:17 AM, Seven Du<dujinfang at gmail.com> wrote:
> Hello All -
> ?? In the spirit of ClueCon (which we are missing this year, but hopefully
> not next), we wanted to document our "FreeSWITCH Story". ?We've posted it to
> the
> wiki(http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
> and it is copied below.
> Thank you all and enjoy a good conference!
> Seven Du (seven)
> Jonathan Palley (jpalley_idapted)
> Idapted Ltd.
>
> How FreeSWITCH has created hundreds of job opportunities and changed lives.
> We want to share our experience working with FreeSWITCH. ?FreeSWITCH has
> been a key enabler of our business. ?We hope this story can be a small way
> to say a very big THANK YOU ALL.
> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
> really allowed us to do just that.
> What We Do:
> We are not a telephony business; we are an educational technology and
> service business. In Asia (China, in our case) students must pass English
> examinations to study or work abroad and gain new experiences. ?However,
> there is limited access to native English speakers and the access students
> can gain is typically very expensive. ?At the same time, in the U.S., there
> are many professionals looking for work-at-home opportunities - people who
> need jobs and would create great teachers. ?Through our technology and
> content we empower these people to be effective English teachers. ?Does it
> work? ?Yes. ?The majority of our students are getting test scores that many
> failed for years to get. ?Just hours ago one student called one of our sales
> agents crying with joy. ?And for our teachers, they are now working in an
> industry that was previously unavailable to those living in the U.S.
> ?http://www.idapted.com
> Why FreeSWITCH Enables This:
> FreeSWITCH has been a key enabler of our business. ?Recording calls,
> controlling routing, integrating with various web-based interfaces, enabling
> multiple endpoints - these are all key features of what we must do. ?Most
> importantly, setting up various servers and routes to mitigate cross-Pacific
> and country-specific network challenges is key. ?Doing what we are doing
> with commercial solutions would have made the business unworkable.
> Our Experiences with FreeSWITCH:
> We started using FreeSWITCH as our VoIP Platform in April 2008, after
> receiving unsatisfactory results with other open source solutions. ?It took
> one day of reading through the FreeSWITCH source code to know, "this is it.
> ?This is the VoIP platform we build our business on". ?It took a few days of
> working with the extremely competent and focused community to re-affirm this
> commitment.
> Our Setup:
> Our teachers use a custom software that integrates a VoIP client with our
> web based platform. Students connect to our teachers "on-demand". ?Simply
> put, on a web-based comet interface the student enters a phone number (or a
> skype name or a gtalk account) and our platform bridges the best available
> trainer and the student. ?At the same time a web-based interface is being
> updated.
> The challenge for us is the connection between teachers and students over a
> cross-continent network. For example, we experienced problems earlier this
> year when a Asis-Pacific communication fiber broken... So, we've learned to
> setup multi servers in multiple datacenters for redundancy.
>
> We run multi instances of FreeSWITCH so we can always use the cutting edge
> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
> only loads mod_dingaling. Here is one beauty of FS: We just had to create
> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
> /usr/local/gtalk etc). This allows us to run the same code base over
> different configurations, and call skype and gtalk accounts just like a
> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
> abnormally or crashes, we can easily change to another FS-skype server (we
> run other servers located in various places in China and HK for
> redundancy).
> FS --|
> ?? ? |---PSTN gateways
> ?? ? |--- FS-skype
> ?? ? |--- FS-gtalk
> ?? ? |--- FS-skype2
> ?? ? |--- more ...
>
>
> COMMUNITY:
> The community's commitment cannot be undervalued. ?The insightful, modular
> design of FreeSWITCH allows anyone to contribute, whereever their skills
> lie. ?It also allows us to easily make modifications to the underlying code
> to suit our specific use-cases ?We want to highlight a few key people and
> modules in the FS ecosystem:
> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
> clients. ?PSTN is zero-conf for the user and mitigates troubles with the end
> users network/microphone, etc (which is significant with our user base).
> ?However, cheap providers fail randomly and FreeSWITCH's ability to control
> routing, use multiple endpoints all while clearly seeing what is going on is
> key.
> Most importantly, anthm and the core team have been super helpful in getting
> SIP to work with us. ?Back in the pre 1.0 days anthm made significant
> changes to mod-sofia to enable clients behind nats without STUN. ?Its
> important to point out that he didn't just make the changes -he forced us to
> really make a compelling case as to why the changes were important for
> FreeSWITCH. ?This is a good thing.
> skype (mod_skypiax): Due to the facts that users prefer skype, we configured
> skypiax. It was unstable at the beginning and that's one of the reason we
> started running that separate FS instance. ?To be fair, it has caused a lot
> of trouble - but we know this, its new software that takes a big risk and
> implements a complex hack. ?What is important is that the author of
> skypiax(Giovanni Maruzzelli) has been a huge help. He's been very active
> fixing bugs and logging in to our box to help trouble shoot. We owe him a
> *big* thanks.
> To make Skypiax more useful, we also created some patches including the ANY
> and RR interfaces for sequential and round robin line hunting, some bug
> fixes and other features like continue-load-on-fail and auto-skype-user
> which haven't been merged into trunk yet. Thanks a community that gives us a
> platform where we can all benefit and contribute.
> erlang (mod_erlang_events): Another key enabler of the next release of our
> system is the erlang interface. ?We have a complex realtime queue routing
> system has it handles input not just from freeswitch, but numerous other web
> interfaces and sockets. ?Erlang was the perfect technology to implement this
> in and luckily an Erlang module for FreeSWITCH was already written.
> Beautiful.
> THE MORAL OF THE STORY:
> FreeSWITCH is a great piece of software that has enabled new technologies
> and business models. ?The design has allowed (and the core team has
> nurtured) a vibrant and exciting community that has made the software even
> better. ?Every day we go to work excited to push the boundaries of what can
> be done with telephony technology and are confident this is the platform of
> the future.
> Thank you all.
>
> Sincerely,
> Du Jinfang (Seven) - Technical Operations/VoIP Manager
> Jonathan Palley - CTO
> Idapted Ltd.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From andrew at hijacked.us  Tue Aug  4 06:49:27 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 4 Aug 2009 09:49:27 -0400
Subject: [Freeswitch-users] event socket vs erlang
In-Reply-To: <20ad6b920908040408x48802b84mbf83e20a2cb5d2f@mail.gmail.com>
References: <20ad6b920908040408x48802b84mbf83e20a2cb5d2f@mail.gmail.com>
Message-ID: <20090804134926.GA27629@hijacked.us>

On Tue, Aug 04, 2009 at 07:08:36PM +0800, mark morreny wrote:
> Hi,
> 
> I have seen people using both event socket and erlang to control freeSWITCH
> externally.
> 
> What is the pros and cons of using event socket vs erlang?
>

It depends on if you want to use erlang or not. The erlang module
provides most of the event socket functionality plus a couple extras
(dynamic XML bindings ala xml_curl, intelligent message delivery, etc).
If you're not already planning to use erlang, it's probably better to
dig out the relevant event socket library module for your langague
instead.

Andrew - author of the erlang module


From jmesquita at gmail.com  Tue Aug  4 07:04:03 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 4 Aug 2009 11:04:03 -0300
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
Message-ID: <5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>

If this is good for me to hear, I would imagine to the core team.

Despite of this not being a group support meeting, I have to say that: Thank
you for sharing, Seven.

jmesquita

On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:

> Hello All -   In the spirit of ClueCon (which we are missing this year,
> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>  We've posted it to the wiki(
> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com) and
> it is copied below.
>
> Thank you all and enjoy a good conference!
>
> Seven Du (seven)
> Jonathan Palley (jpalley_idapted)
> Idapted Ltd.
>
>
> *How FreeSWITCH has created hundreds of job opportunities and changed
> lives. *
>
> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
> been a key enabler of our business.  We hope this story can be a small way
> to say a very big THANK YOU ALL.
>
> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
> really allowed us to do just that.
>
> What We Do:
> We are not a telephony business; we are an educational technology and
> service business. In Asia (China, in our case) students must pass English
> examinations to study or work abroad and gain new experiences.  However,
> there is limited access to native English speakers and the access students
> can gain is typically very expensive.  At the same time, in the U.S., there
> are many professionals looking for work-at-home opportunities - people who
> need jobs and would create great teachers.  Through our technology and
> content we empower these people to be effective English teachers.  Does it
> work?  Yes.  The majority of our students are getting test scores that many
> failed for years to get.  Just hours ago one student called one of our sales
> agents crying with joy.  And for our teachers, they are now working in an
> industry that was previously unavailable to those living in the U.S.
> http://www.idapted.com
>
> Why FreeSWITCH Enables This:
> FreeSWITCH has been a key enabler of our business.  Recording calls,
> controlling routing, integrating with various web-based interfaces, enabling
> multiple endpoints - these are all key features of what we must do.  Most
> importantly, setting up various servers and routes to mitigate cross-Pacific
> and country-specific network challenges is key.  Doing what we are doing
> with commercial solutions would have made the business unworkable.
>
> Our Experiences with FreeSWITCH:
> We started using FreeSWITCH as our VoIP Platform in April 2008, after
> receiving unsatisfactory results with other open source solutions.  It took
> one day of reading through the FreeSWITCH source code to know, "this is it.
>  This is the VoIP platform we build our business on".  It took a few days of
> working with the extremely competent and focused community to re-affirm this
> commitment.
>
> Our Setup:
> Our teachers use a custom software that integrates a VoIP client with our
> web based platform. Students connect to our teachers "on-demand".  Simply
> put, on a web-based comet interface the student enters a phone number (or a
> skype name or a gtalk account) and our platform bridges the best available
> trainer and the student.  At the same time a web-based interface is being
> updated.
>
> The challenge for us is the connection between teachers and students over a
> cross-continent network. For example, we experienced problems earlier this
> year when a Asis-Pacific communication fiber broken... So, we've learned to
> setup multi servers in multiple datacenters for redundancy.
>
> We run multi instances of FreeSWITCH so we can always use the cutting edge
> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
> only loads mod_dingaling. Here is one beauty of FS: We just had to create
> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
> /usr/local/gtalk etc). This allows us to run the same code base over
> different configurations, and call skype and gtalk accounts just like a
> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
> abnormally or crashes, we can easily change to another FS-skype server (we
> run other servers located in various places in China and HK for
> redundancy).
>
> FS --|
>      |---PSTN gateways
>      |--- FS-skype
>      |--- FS-gtalk
>      |--- FS-skype2
>      |--- more ...
>
>
>
> COMMUNITY:
>
> The community's commitment cannot be undervalued.  The insightful, modular
> design of FreeSWITCH allows anyone to contribute, whereever their skills
> lie.  It also allows us to easily make modifications to the underlying code
> to suit our specific use-cases  We want to highlight a few key people and
> modules in the FS ecosystem:
>
> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
> users network/microphone, etc (which is significant with our user base).
>  However, cheap providers fail randomly and FreeSWITCH's ability to control
> routing, use multiple endpoints all while clearly seeing what is going on is
> key.
> Most importantly, anthm and the core team have been super helpful in
> getting SIP to work with us.  Back in the pre 1.0 days anthm made
> significant changes to mod-sofia to enable clients behind nats without STUN.
>  Its important to point out that he didn't just make the changes -he forced
> us to really make a compelling case as to why the changes were important for
> FreeSWITCH.  This is a good thing.
>
> skype (mod_skypiax): Due to the facts that users prefer skype, we
> configured skypiax. It was unstable at the beginning and that's one of the
> reason we started running that separate FS instance.  To be fair, it has
> caused a lot of trouble - but we know this, its new software that takes a
> big risk and implements a complex hack.  What is important is that the
> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
> active fixing bugs and logging in to our box to help trouble shoot. We owe
> him a *big* thanks.
>
> To make Skypiax more useful, we also created some patches including the ANY
> and RR interfaces for sequential and round robin line hunting, some bug
> fixes and other features like continue-load-on-fail and auto-skype-user
> which haven't been merged into trunk yet. Thanks a community that gives us a
> platform where we can all benefit and contribute.
>
> erlang (mod_erlang_events): Another key enabler of the next release of our
> system is the erlang interface.  We have a complex realtime queue routing
> system has it handles input not just from freeswitch, but numerous other web
> interfaces and sockets.  Erlang was the perfect technology to implement this
> in and luckily an Erlang module for FreeSWITCH was already written.
> Beautiful.
>
> THE MORAL OF THE STORY:
> FreeSWITCH is a great piece of software that has enabled new technologies
> and business models.  The design has allowed (and the core team has
> nurtured) a vibrant and exciting community that has made the software even
> better.  Every day we go to work excited to push the boundaries of what can
> be done with telephony technology and are confident this is the platform of
> the future.
>
> Thank you all.
>
>
> Sincerely,
>
> Du Jinfang (Seven) - Technical Operations/VoIP Manager
> Jonathan Palley - CTO
> Idapted Ltd.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rdenert at tng.de  Tue Aug  4 08:41:18 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Tue, 4 Aug 2009 17:41:18 +0200 (CEST)
Subject: [Freeswitch-users] Module in Lua not working
In-Reply-To: <7203453.221221249384483704.JavaMail.root@zimbra.tng.de>
Message-ID: <17585545.224751249400478914.JavaMail.root@zimbra.tng.de>

Hello,

there is no more problem. I installed 1.0.4 and everthing is fine. :-)

BR

----- Urspr?ngliche Mail -----
Von: "Rudolf Denert" <rdenert at tng.de>
An: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
Gesendet: Dienstag, 4. August 2009 13:14:43 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: [Freeswitch-users] Module in Lua not working

Hello again!

I need some help again, because I have little trouble with a few modules. The first one is luasocket, the seconde one is luasql. I always get the error:

error loading module 'socket' from file '/usr/local/lib/lua/5.1/socket/core.so': /usr/local/lib/lua/5.1/socket/core.so: undefined symbol: lua_getmetatable

and

error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_pushlstring

in the fs_cli

This looks like a conflict with an older version of these modules. Is this right? If yes, what should i update?

Thanks again.

BR

-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
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-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
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Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.


From matt at hellohunter.com  Tue Aug  4 11:50:36 2009
From: matt at hellohunter.com (Matt Hunter)
Date: Tue, 4 Aug 2009 11:50:36 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
Message-ID: <4256bf830908041150x5a450797j62389d4513997f9c@mail.gmail.com>

Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
logs below, but I am still at a loss at being able to identify the error or
reproduce it consistently. The below log indicates to me that my FS server
is initiating sending 2 BYE message to my DID provider (didforsale.com). Is
there a way I can look further inside FreeSWITCH to see why it is sending
this BYE packet?

sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0


sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0



On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> Digging a bit in mod_sofia releaved that it can be caused by a SIP
> code 482 (loop detected), 483 (too many hops) or 484 (address
> incomplete).
>
> Do a SIP trace to sched more light on what's happening.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong:
>
> > EXCHANGE_ROUTING_ERROR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mattdfong at gmail.com  Tue Aug  4 11:51:03 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 4 Aug 2009 11:51:03 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
Message-ID: <4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>

Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
logs below, but I am still at a loss at being able to identify the error or
reproduce it consistently. The below log indicates to me that my FS server
is initiating sending 2 BYE message to my DID provider (didforsale.com). Is
there a way I can look further inside FreeSWITCH to see why it is sending
this BYE packet?

sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0


sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0

On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> Digging a bit in mod_sofia releaved that it can be caused by a SIP
> code 482 (loop detected), 483 (too many hops) or 484 (address
> incomplete).
>
> Do a SIP trace to sched more light on what's happening.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong:
>
> > EXCHANGE_ROUTING_ERROR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gregt at cgicommunications.com  Tue Aug  4 12:12:25 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Tue, 4 Aug 2009 15:12:25 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
Message-ID: <E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>

Hi, does anyone have an example of a simple PHP socket script that  
will listen and spawn of a process that handles the incoming call?

I understand the inbound socket and code such as this
http://wiki.freeswitch.org/wiki/PHP_Event_Socket
that will let me initiate operations. It's the constantly running php  
socket program that I can't get my head around, and how it will spawn  
another php script that will be able to do things like answer the  
session, get dtmf, etc.
--
Greg


On Aug 2, 2009, at 6:35 PM, Michael Collins wrote:

>
>
> On Sun, Aug 2, 2009 at 12:38 PM, Nik Middleton <nik.middleton at noblesolutions.co.uk 
> > wrote:
> Hi Guys,
>
>
> I?m using an outbound socket to control calls, and it works a  
> charm.  However, what I?d like to do is send a custom event  
> regarding the call on hang-up.  The way I see things happening at  
> the moment, and I could be wrong, is that the socket is closed when  
> a hang-up occurs, so am I taking a chance trying to send the event  
> then? (try to sneak out the event before socket closure happens)   
> The other option is of course to open an inbound socket and send the  
> event, but I?d rather not do that if possible.
>
> Nik,
>
> Perhaps the "linger" event socket command will do what you need?  
> Check out this commit:
> http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009391.html
>
> Let me know if it works for you and I'll be sure to get it  
> documented properly. If you get it working I'd love to see a code  
> snippet so we can wikify this knowledge. :)
>
> Thanks,
> MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Tue Aug  4 16:04:08 2009
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 4 Aug 2009 19:04:08 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local> 
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com> 
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
Message-ID: <6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>

I wrote some notes on this but have yet to wiki it.

example of an outbound socket connection where the call is answered, a
variable is set then perhaps play one of the pre-installed files and
hangup.
ivrd

fs_ivrd comes with freeswitch. It being a small daemon just invokes
the script defined in a variable and passes data from it via STDIN/OUT

Since this is an outbound socket connections it needs to be defined in
the dialplan.



Ex:

<extension name="outbound-socket">

    <condition field="destination_number" expression="^55(522)$">

        <action application="set"
data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>

        <action application="socket" data="127.0.0.1:8084 async full"/>

    </condition>

  </extension>



The above dialplan sample would invoke ivr-demo.php when 55522 is
called as long as fs_ivrd is running. To start fs_ivrd:

/usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8004



It takes 2 arguments -h for hostname and -p for port.



PHP Code

#!/usr/bin/php -q

<?php



    // set a couple of things so we dont kill the system

    ob_implicit_flush(true);

    set_time_limit(30);





    // Open stdin so we can read the AGI data in

    $in = fopen("php://stdin", "r");

    // Connect

    echo "connect\n\n";

    // Answer

    echo "sendmsg\n";

    echo "call-command: execute\n";

    echo "execute-app-name: answer\n\n";



    // Play a prompt

    echo "sendmsg\n";

    echo "call-command: execute\n";

    echo "execute-app-name: playback\n";

    echo "execute-app-arg:
/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n";



    // Wait

    sleep(5);



   // Hangup

   echo "sendmsg\n";

   echo "call-command: hangup\n\n";



fclose($in);



?>



ivrd will call this script for each call. All itdoes is answer the
channel tell FreeSWITCH to play the ?welcome to freeswitch? prompt.
Since the script is now controlling all call flow I needed to add a
wait or it would send the hangup immediately before the prompt was
played.

Some improvements possible but that's 1 way to do it. It would be
possible to do the socket directly in PHP but fs_ivrd is a nice option
too.

-- W


From diego.viola at gmail.com  Tue Aug  4 17:54:34 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 20:54:34 -0400
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
Message-ID: <86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>

Very cool, and yes, FreeSWITCH does rock =D

Both the software and the community ;)

2009/8/4 Jo?o Mesquita <jmesquita at gmail.com>

> If this is good for me to hear, I would imagine to the core team.
>
> Despite of this not being a group support meeting, I have to say that:
> Thank you for sharing, Seven.
>
> jmesquita
>
> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:
>
>> Hello All -   In the spirit of ClueCon (which we are missing this year,
>> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>  We've posted it to the wiki(
>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>> and it is copied below.
>>
>> Thank you all and enjoy a good conference!
>>
>> Seven Du (seven)
>> Jonathan Palley (jpalley_idapted)
>> Idapted Ltd.
>>
>>
>> *How FreeSWITCH has created hundreds of job opportunities and changed
>> lives. *
>>
>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>> been a key enabler of our business.  We hope this story can be a small way
>> to say a very big THANK YOU ALL.
>>
>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
>> really allowed us to do just that.
>>
>> What We Do:
>> We are not a telephony business; we are an educational technology and
>> service business. In Asia (China, in our case) students must pass English
>> examinations to study or work abroad and gain new experiences.  However,
>> there is limited access to native English speakers and the access students
>> can gain is typically very expensive.  At the same time, in the U.S., there
>> are many professionals looking for work-at-home opportunities - people who
>> need jobs and would create great teachers.  Through our technology and
>> content we empower these people to be effective English teachers.  Does it
>> work?  Yes.  The majority of our students are getting test scores that many
>> failed for years to get.  Just hours ago one student called one of our sales
>> agents crying with joy.  And for our teachers, they are now working in an
>> industry that was previously unavailable to those living in the U.S.
>> http://www.idapted.com
>>
>> Why FreeSWITCH Enables This:
>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>> controlling routing, integrating with various web-based interfaces, enabling
>> multiple endpoints - these are all key features of what we must do.  Most
>> importantly, setting up various servers and routes to mitigate cross-Pacific
>> and country-specific network challenges is key.  Doing what we are doing
>> with commercial solutions would have made the business unworkable.
>>
>> Our Experiences with FreeSWITCH:
>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>> receiving unsatisfactory results with other open source solutions.  It took
>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>  This is the VoIP platform we build our business on".  It took a few days of
>> working with the extremely competent and focused community to re-affirm this
>> commitment.
>>
>> Our Setup:
>> Our teachers use a custom software that integrates a VoIP client with our
>> web based platform. Students connect to our teachers "on-demand".  Simply
>> put, on a web-based comet interface the student enters a phone number (or a
>> skype name or a gtalk account) and our platform bridges the best available
>> trainer and the student.  At the same time a web-based interface is being
>> updated.
>>
>> The challenge for us is the connection between teachers and students over
>> a cross-continent network. For example, we experienced problems earlier this
>> year when a Asis-Pacific communication fiber broken... So, we've learned to
>> setup multi servers in multiple datacenters for redundancy.
>>
>> We run multi instances of FreeSWITCH so we can always use the cutting edge
>> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
>> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
>> only loads mod_dingaling. Here is one beauty of FS: We just had to create
>> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>> /usr/local/gtalk etc). This allows us to run the same code base over
>> different configurations, and call skype and gtalk accounts just like a
>> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
>> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
>> abnormally or crashes, we can easily change to another FS-skype server (we
>> run other servers located in various places in China and HK for
>> redundancy).
>>
>> FS --|
>>      |---PSTN gateways
>>      |--- FS-skype
>>      |--- FS-gtalk
>>      |--- FS-skype2
>>      |--- more ...
>>
>>
>>
>> COMMUNITY:
>>
>> The community's commitment cannot be undervalued.  The insightful, modular
>> design of FreeSWITCH allows anyone to contribute, whereever their skills
>> lie.  It also allows us to easily make modifications to the underlying code
>> to suit our specific use-cases  We want to highlight a few key people and
>> modules in the FS ecosystem:
>>
>> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
>> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
>> users network/microphone, etc (which is significant with our user base).
>>  However, cheap providers fail randomly and FreeSWITCH's ability to control
>> routing, use multiple endpoints all while clearly seeing what is going on is
>> key.
>> Most importantly, anthm and the core team have been super helpful in
>> getting SIP to work with us.  Back in the pre 1.0 days anthm made
>> significant changes to mod-sofia to enable clients behind nats without STUN.
>>  Its important to point out that he didn't just make the changes -he forced
>> us to really make a compelling case as to why the changes were important for
>> FreeSWITCH.  This is a good thing.
>>
>> skype (mod_skypiax): Due to the facts that users prefer skype, we
>> configured skypiax. It was unstable at the beginning and that's one of the
>> reason we started running that separate FS instance.  To be fair, it has
>> caused a lot of trouble - but we know this, its new software that takes a
>> big risk and implements a complex hack.  What is important is that the
>> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
>> active fixing bugs and logging in to our box to help trouble shoot. We owe
>> him a *big* thanks.
>>
>> To make Skypiax more useful, we also created some patches including the
>> ANY and RR interfaces for sequential and round robin line hunting, some bug
>> fixes and other features like continue-load-on-fail and auto-skype-user
>> which haven't been merged into trunk yet. Thanks a community that gives us a
>> platform where we can all benefit and contribute.
>>
>> erlang (mod_erlang_events): Another key enabler of the next release of our
>> system is the erlang interface.  We have a complex realtime queue routing
>> system has it handles input not just from freeswitch, but numerous other web
>> interfaces and sockets.  Erlang was the perfect technology to implement this
>> in and luckily an Erlang module for FreeSWITCH was already written.
>> Beautiful.
>>
>> THE MORAL OF THE STORY:
>> FreeSWITCH is a great piece of software that has enabled new technologies
>> and business models.  The design has allowed (and the core team has
>> nurtured) a vibrant and exciting community that has made the software even
>> better.  Every day we go to work excited to push the boundaries of what can
>> be done with telephony technology and are confident this is the platform of
>> the future.
>>
>> Thank you all.
>>
>>
>> Sincerely,
>>
>> Du Jinfang (Seven) - Technical Operations/VoIP Manager
>> Jonathan Palley - CTO
>> Idapted Ltd.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Tue Aug  4 17:55:22 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 20:55:22 -0400
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
Message-ID: <86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>

Cool, I wish I could be there, next year =D

Any pics or videos of ClueCon? :)

On Mon, Aug 3, 2009 at 8:41 AM, Brian West <brian at freeswitch.org> wrote:

> Just look for large groups of people with laptops.  I'm sure you can't
> miss us.
>
> /b
>
> On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
>
> > i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> > afternoon?  anybody need help with setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Tue Aug  4 17:56:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 20:56:27 -0400
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
Message-ID: <86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>

Maybe you can link your testimonial or put it here also? :D

http://wiki.freeswitch.org/wiki/Testimonials

On Tue, Aug 4, 2009 at 8:54 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Very cool, and yes, FreeSWITCH does rock =D
>
> Both the software and the community ;)
>
> 2009/8/4 Jo?o Mesquita <jmesquita at gmail.com>
>
> If this is good for me to hear, I would imagine to the core team.
>>
>> Despite of this not being a group support meeting, I have to say that:
>> Thank you for sharing, Seven.
>>
>> jmesquita
>>
>> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>>>  Hello All -   In the spirit of ClueCon (which we are missing this year,
>>> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>>  We've posted it to the wiki(
>>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>>> and it is copied below.
>>>
>>> Thank you all and enjoy a good conference!
>>>
>>> Seven Du (seven)
>>> Jonathan Palley (jpalley_idapted)
>>> Idapted Ltd.
>>>
>>>
>>> *How FreeSWITCH has created hundreds of job opportunities and changed
>>> lives. *
>>>
>>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>>> been a key enabler of our business.  We hope this story can be a small way
>>> to say a very big THANK YOU ALL.
>>>
>>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
>>> really allowed us to do just that.
>>>
>>> What We Do:
>>> We are not a telephony business; we are an educational technology and
>>> service business. In Asia (China, in our case) students must pass English
>>> examinations to study or work abroad and gain new experiences.  However,
>>> there is limited access to native English speakers and the access students
>>> can gain is typically very expensive.  At the same time, in the U.S., there
>>> are many professionals looking for work-at-home opportunities - people who
>>> need jobs and would create great teachers.  Through our technology and
>>> content we empower these people to be effective English teachers.  Does it
>>> work?  Yes.  The majority of our students are getting test scores that many
>>> failed for years to get.  Just hours ago one student called one of our sales
>>> agents crying with joy.  And for our teachers, they are now working in an
>>> industry that was previously unavailable to those living in the U.S.
>>> http://www.idapted.com
>>>
>>> Why FreeSWITCH Enables This:
>>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>>> controlling routing, integrating with various web-based interfaces, enabling
>>> multiple endpoints - these are all key features of what we must do.  Most
>>> importantly, setting up various servers and routes to mitigate cross-Pacific
>>> and country-specific network challenges is key.  Doing what we are doing
>>> with commercial solutions would have made the business unworkable.
>>>
>>> Our Experiences with FreeSWITCH:
>>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>>> receiving unsatisfactory results with other open source solutions.  It took
>>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>>  This is the VoIP platform we build our business on".  It took a few days of
>>> working with the extremely competent and focused community to re-affirm this
>>> commitment.
>>>
>>> Our Setup:
>>> Our teachers use a custom software that integrates a VoIP client with our
>>> web based platform. Students connect to our teachers "on-demand".  Simply
>>> put, on a web-based comet interface the student enters a phone number (or a
>>> skype name or a gtalk account) and our platform bridges the best available
>>> trainer and the student.  At the same time a web-based interface is being
>>> updated.
>>>
>>> The challenge for us is the connection between teachers and students over
>>> a cross-continent network. For example, we experienced problems earlier this
>>> year when a Asis-Pacific communication fiber broken... So, we've learned to
>>> setup multi servers in multiple datacenters for redundancy.
>>>
>>> We run multi instances of FreeSWITCH so we can always use the cutting
>>> edge and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS)
>>> instance connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and
>>> FS-gtalk only loads mod_dingaling. Here is one beauty of FS: We just had to
>>> create different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>>> /usr/local/gtalk etc). This allows us to run the same code base over
>>> different configurations, and call skype and gtalk accounts just like a
>>> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
>>> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
>>> abnormally or crashes, we can easily change to another FS-skype server (we
>>> run other servers located in various places in China and HK for
>>> redundancy).
>>>
>>> FS --|
>>>      |---PSTN gateways
>>>      |--- FS-skype
>>>      |--- FS-gtalk
>>>      |--- FS-skype2
>>>      |--- more ...
>>>
>>>
>>>
>>> COMMUNITY:
>>>
>>> The community's commitment cannot be undervalued.  The insightful,
>>> modular design of FreeSWITCH allows anyone to contribute, whereever their
>>> skills lie.  It also allows us to easily make modifications to the
>>> underlying code to suit our specific use-cases  We want to highlight a few
>>> key people and modules in the FS ecosystem:
>>>
>>> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
>>> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
>>> users network/microphone, etc (which is significant with our user base).
>>>  However, cheap providers fail randomly and FreeSWITCH's ability to control
>>> routing, use multiple endpoints all while clearly seeing what is going on is
>>> key.
>>> Most importantly, anthm and the core team have been super helpful in
>>> getting SIP to work with us.  Back in the pre 1.0 days anthm made
>>> significant changes to mod-sofia to enable clients behind nats without STUN.
>>>  Its important to point out that he didn't just make the changes -he forced
>>> us to really make a compelling case as to why the changes were important for
>>> FreeSWITCH.  This is a good thing.
>>>
>>> skype (mod_skypiax): Due to the facts that users prefer skype, we
>>> configured skypiax. It was unstable at the beginning and that's one of the
>>> reason we started running that separate FS instance.  To be fair, it has
>>> caused a lot of trouble - but we know this, its new software that takes a
>>> big risk and implements a complex hack.  What is important is that the
>>> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
>>> active fixing bugs and logging in to our box to help trouble shoot. We owe
>>> him a *big* thanks.
>>>
>>> To make Skypiax more useful, we also created some patches including the
>>> ANY and RR interfaces for sequential and round robin line hunting, some bug
>>> fixes and other features like continue-load-on-fail and auto-skype-user
>>> which haven't been merged into trunk yet. Thanks a community that gives us a
>>> platform where we can all benefit and contribute.
>>>
>>> erlang (mod_erlang_events): Another key enabler of the next release of
>>> our system is the erlang interface.  We have a complex realtime queue
>>> routing system has it handles input not just from freeswitch, but numerous
>>> other web interfaces and sockets.  Erlang was the perfect technology to
>>> implement this in and luckily an Erlang module for FreeSWITCH was already
>>> written. Beautiful.
>>>
>>> THE MORAL OF THE STORY:
>>> FreeSWITCH is a great piece of software that has enabled new technologies
>>> and business models.  The design has allowed (and the core team has
>>> nurtured) a vibrant and exciting community that has made the software even
>>> better.  Every day we go to work excited to push the boundaries of what can
>>> be done with telephony technology and are confident this is the platform of
>>> the future.
>>>
>>> Thank you all.
>>>
>>>
>>> Sincerely,
>>>
>>> Du Jinfang (Seven) - Technical Operations/VoIP Manager
>>> Jonathan Palley - CTO
>>> Idapted Ltd.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From gabe at gundy.org  Tue Aug  4 18:53:23 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 4 Aug 2009 19:53:23 -0600
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
	<86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>
Message-ID: <903da5680908041853y7f828878pc539470b328066dc@mail.gmail.com>

On Tue, Aug 4, 2009 at 6:55 PM, Diego Viola<diego.viola at gmail.com> wrote:
> Cool, I wish I could be there, next year =D

Tomorrow is my wedding anniversary 8/5.  If ClueCon keeps getting
scheduled on that date, I don't know if I'll ever get to go :(


> Any pics or videos of ClueCon? :)

Yes, bring on the vids.

Best,
Gabe


From diego.viola at gmail.com  Tue Aug  4 18:56:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 21:56:27 -0400
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <903da5680908041853y7f828878pc539470b328066dc@mail.gmail.com>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
	<86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>
	<903da5680908041853y7f828878pc539470b328066dc@mail.gmail.com>
Message-ID: <86a32abc0908041856u4d4a5c4byff3c9f73be0e5031@mail.gmail.com>

Looking forward to this talk =D

"FreeSWITCH: Learning to Think Fourth Dimensionally"

;)

On Tue, Aug 4, 2009 at 9:53 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Tue, Aug 4, 2009 at 6:55 PM, Diego Viola<diego.viola at gmail.com> wrote:
> > Cool, I wish I could be there, next year =D
>
> Tomorrow is my wedding anniversary 8/5.  If ClueCon keeps getting
> scheduled on that date, I don't know if I'll ever get to go :(
>
>
> > Any pics or videos of ClueCon? :)
>
> Yes, bring on the vids.
>
> Best,
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Tue Aug  4 19:22:07 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 22:22:07 -0400
Subject: [Freeswitch-users] Looking for some FreeSWITCH job
Message-ID: <86a32abc0908041922j5dac756g49b2d87db74479b1@mail.gmail.com>

Hi,

I'm currently looking for some FS jobs, I really need one, I'm currently
unemployed and looking for some serious FreeSWITCH jobs.

Anyone?

P.S: I also do FS and web development with any language, PHP, Ruby, etc.
anything really.

Thanks,

Diego
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From jpalley at idapted.com  Tue Aug  4 21:33:32 2009
From: jpalley at idapted.com (Jonathan Palley)
Date: Wed, 5 Aug 2009 12:33:32 +0800
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com> 
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com> 
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com> 
	<86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
Message-ID: <2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>

Diego -   Already done.  See the bottom of the page (we linked to another
page because of its length)!

:)
Jonathan Palley
Idapted Ltd.

On Wed, Aug 5, 2009 at 8:56 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Maybe you can link your testimonial or put it here also? :D
>
> http://wiki.freeswitch.org/wiki/Testimonials
>
>
> On Tue, Aug 4, 2009 at 8:54 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Very cool, and yes, FreeSWITCH does rock =D
>>
>> Both the software and the community ;)
>>
>> 2009/8/4 Jo?o Mesquita <jmesquita at gmail.com>
>>
>> If this is good for me to hear, I would imagine to the core team.
>>>
>>> Despite of this not being a group support meeting, I have to say that:
>>> Thank you for sharing, Seven.
>>>
>>> jmesquita
>>>
>>> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>>>  Hello All -   In the spirit of ClueCon (which we are missing this
>>>> year, but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>>>  We've posted it to the wiki(
>>>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>>>> and it is copied below.
>>>>
>>>> Thank you all and enjoy a good conference!
>>>>
>>>> Seven Du (seven)
>>>> Jonathan Palley (jpalley_idapted)
>>>> Idapted Ltd.
>>>>
>>>>
>>>> *How FreeSWITCH has created hundreds of job opportunities and changed
>>>> lives. *
>>>>
>>>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>>>> been a key enabler of our business.  We hope this story can be a small way
>>>> to say a very big THANK YOU ALL.
>>>>
>>>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH
>>>> has really allowed us to do just that.
>>>>
>>>> What We Do:
>>>> We are not a telephony business; we are an educational technology and
>>>> service business. In Asia (China, in our case) students must pass English
>>>> examinations to study or work abroad and gain new experiences.  However,
>>>> there is limited access to native English speakers and the access students
>>>> can gain is typically very expensive.  At the same time, in the U.S., there
>>>> are many professionals looking for work-at-home opportunities - people who
>>>> need jobs and would create great teachers.  Through our technology and
>>>> content we empower these people to be effective English teachers.  Does it
>>>> work?  Yes.  The majority of our students are getting test scores that many
>>>> failed for years to get.  Just hours ago one student called one of our sales
>>>> agents crying with joy.  And for our teachers, they are now working in an
>>>> industry that was previously unavailable to those living in the U.S.
>>>> http://www.idapted.com
>>>>
>>>> Why FreeSWITCH Enables This:
>>>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>>>> controlling routing, integrating with various web-based interfaces, enabling
>>>> multiple endpoints - these are all key features of what we must do.  Most
>>>> importantly, setting up various servers and routes to mitigate cross-Pacific
>>>> and country-specific network challenges is key.  Doing what we are doing
>>>> with commercial solutions would have made the business unworkable.
>>>>
>>>> Our Experiences with FreeSWITCH:
>>>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>>>> receiving unsatisfactory results with other open source solutions.  It took
>>>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>>>  This is the VoIP platform we build our business on".  It took a few days of
>>>> working with the extremely competent and focused community to re-affirm this
>>>> commitment.
>>>>
>>>> Our Setup:
>>>> Our teachers use a custom software that integrates a VoIP client with
>>>> our web based platform. Students connect to our teachers "on-demand".
>>>>  Simply put, on a web-based comet interface the student enters a phone
>>>> number (or a skype name or a gtalk account) and our platform bridges the
>>>> best available trainer and the student.  At the same time a web-based
>>>> interface is being updated.
>>>>
>>>> The challenge for us is the connection between teachers and students
>>>> over a cross-continent network. For example, we experienced problems earlier
>>>> this year when a Asis-Pacific communication fiber broken... So, we've
>>>> learned to setup multi servers in multiple datacenters for redundancy.
>>>>
>>>> We run multi instances of FreeSWITCH so we can always use the cutting
>>>> edge and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS)
>>>> instance connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and
>>>> FS-gtalk only loads mod_dingaling. Here is one beauty of FS: We just had to
>>>> create different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>>>> /usr/local/gtalk etc). This allows us to run the same code base over
>>>> different configurations, and call skype and gtalk accounts just like a
>>>> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
>>>> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
>>>> abnormally or crashes, we can easily change to another FS-skype server (we
>>>> run other servers located in various places in China and HK for
>>>> redundancy).
>>>>
>>>> FS --|
>>>>      |---PSTN gateways
>>>>      |--- FS-skype
>>>>      |--- FS-gtalk
>>>>      |--- FS-skype2
>>>>      |--- more ...
>>>>
>>>>
>>>>
>>>> COMMUNITY:
>>>>
>>>> The community's commitment cannot be undervalued.  The insightful,
>>>> modular design of FreeSWITCH allows anyone to contribute, whereever their
>>>> skills lie.  It also allows us to easily make modifications to the
>>>> underlying code to suit our specific use-cases  We want to highlight a few
>>>> key people and modules in the FS ecosystem:
>>>>
>>>> mod_sofia: SIP is how we connect to our PSTN gateways and to our
>>>> teachers clients.  PSTN is zero-conf for the user and mitigates troubles
>>>> with the end users network/microphone, etc (which is significant with our
>>>> user base).  However, cheap providers fail randomly and FreeSWITCH's ability
>>>> to control routing, use multiple endpoints all while clearly seeing what is
>>>> going on is key.
>>>> Most importantly, anthm and the core team have been super helpful in
>>>> getting SIP to work with us.  Back in the pre 1.0 days anthm made
>>>> significant changes to mod-sofia to enable clients behind nats without STUN.
>>>>  Its important to point out that he didn't just make the changes -he forced
>>>> us to really make a compelling case as to why the changes were important for
>>>> FreeSWITCH.  This is a good thing.
>>>>
>>>> skype (mod_skypiax): Due to the facts that users prefer skype, we
>>>> configured skypiax. It was unstable at the beginning and that's one of the
>>>> reason we started running that separate FS instance.  To be fair, it has
>>>> caused a lot of trouble - but we know this, its new software that takes a
>>>> big risk and implements a complex hack.  What is important is that the
>>>> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
>>>> active fixing bugs and logging in to our box to help trouble shoot. We owe
>>>> him a *big* thanks.
>>>>
>>>> To make Skypiax more useful, we also created some patches including the
>>>> ANY and RR interfaces for sequential and round robin line hunting, some bug
>>>> fixes and other features like continue-load-on-fail and auto-skype-user
>>>> which haven't been merged into trunk yet. Thanks a community that gives us a
>>>> platform where we can all benefit and contribute.
>>>>
>>>> erlang (mod_erlang_events): Another key enabler of the next release of
>>>> our system is the erlang interface.  We have a complex realtime queue
>>>> routing system has it handles input not just from freeswitch, but numerous
>>>> other web interfaces and sockets.  Erlang was the perfect technology to
>>>> implement this in and luckily an Erlang module for FreeSWITCH was already
>>>> written. Beautiful.
>>>>
>>>> THE MORAL OF THE STORY:
>>>> FreeSWITCH is a great piece of software that has enabled new
>>>> technologies and business models.  The design has allowed (and the core team
>>>> has nurtured) a vibrant and exciting community that has made the software
>>>> even better.  Every day we go to work excited to push the boundaries of what
>>>> can be done with telephony technology and are confident this is the platform
>>>> of the future.
>>>>
>>>> Thank you all.
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Du Jinfang (Seven) - Technical Operations/VoIP Manager
>>>> Jonathan Palley - CTO
>>>> Idapted Ltd.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue Aug  4 21:41:20 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Aug 2009 12:41:20 +0800
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
	<86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
	<2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>
Message-ID: <23f91030908042141h9f79d61w86599122677670@mail.gmail.com>

And I  added this on the wiki page:
mod_conference and mod_fifo: We also use FreeSWITCH in our office
environment as a PBX for call center and customer service connected with
VoIP and PSTN(openzap) gateways. It is integrated into our CRM system
naturally and just made sales process, business logic and world wide
conference much more simpler and easier.

:)

7

2009/8/5 Jonathan Palley <jpalley at idapted.com>

> Diego -   Already done.  See the bottom of the page (we linked to another
> page because of its length)!
>
> :)
> Jonathan Palley
> Idapted Ltd.
>
>
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From diego.viola at gmail.com  Tue Aug  4 22:11:24 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 5 Aug 2009 01:11:24 -0400
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <23f91030908042141h9f79d61w86599122677670@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
	<86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
	<2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>
	<23f91030908042141h9f79d61w86599122677670@mail.gmail.com>
Message-ID: <86a32abc0908042211g2e023d1fgd6f8eb25630561d9@mail.gmail.com>

Very cool, thanks guys, you make FreeSWITCH even better =D

Your story rocks!

On Wed, Aug 5, 2009 at 12:41 AM, Seven Du<dujinfang at gmail.com> wrote:
> And I ?added this on the wiki page:
> mod_conference and mod_fifo: We also use FreeSWITCH in our office
> environment as a PBX for call center and customer service connected with
> VoIP and PSTN(openzap) gateways. It is integrated into our CRM system
> naturally and just made sales process, business logic and world wide
> conference much more simpler and easier.
> :)
> 7
>
> 2009/8/5 Jonathan Palley <jpalley at idapted.com>
>>
>> Diego -
>> ??Already done. ?See the bottom of the page (we linked to another page
>> because of?its?length)!
>> :)
>> Jonathan?Palley
>> Idapted?Ltd.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From velu.technical at gmail.com  Tue Aug  4 23:14:55 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Wed, 5 Aug 2009 11:44:55 +0530
Subject: [Freeswitch-users] execute function in ESL.pm module is not working
Message-ID: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>

Dear All,
   I registered alarm signal in my Perl server program.
   If ALARM signal occurred I execute the following statement in signal
handler.
    "$conn->execute("playback",$sound_path."voicemail/vm-goodbye.wav")"
  The above statement didn't play that wave file. But before generating the
ALARM signal it worked.

What is the problem?

Please help me in this problem....

Also Is there any idea to do timeout for DTMF digits?

Thanks...

Regards,
K.Velusamy
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From brad.tuan at gmail.com  Wed Aug  5 03:26:01 2009
From: brad.tuan at gmail.com (Brad Tuan)
Date: Wed, 5 Aug 2009 18:26:01 +0800
Subject: [Freeswitch-users] How to change the contact when fs sending
	REGISTER??
Message-ID: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>

As title ,I know how to do when sending INVITE

but how to do it when fs sending REGISTER??

For example , when gateway registering , the contact is
gw+abcd at XXX.XXX.XXX.XXX ,

how to change it to *abcd at XXX.XXX.XXX.XXX??*
**
*Please help*
**
**
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From gregt at cgicommunications.com  Wed Aug  5 05:54:04 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 08:54:04 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
	<6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
Message-ID: <FE05AA09-CA4B-40E0-BE43-626667FFEDA9@cgicommunications.com>

Thanks so much, William. This gives me a great start.
--
Greg



On Aug 4, 2009, at 7:04 PM, William Suffill wrote:

> I wrote some notes on this but have yet to wiki it.
>
> example of an outbound socket connection where the call is answered, a
> variable is set then perhaps play one of the pre-installed files and
> hangup.
> ivrd
>
> fs_ivrd comes with freeswitch. It being a small daemon just invokes
> the script defined in a variable and passes data from it via STDIN/OUT
>
> Since this is an outbound socket connections it needs to be defined in
> the dialplan.
>
>
>
> Ex:
>
> <extension name="outbound-socket">
>
>    <condition field="destination_number" expression="^55(522)$">
>
>        <action application="set"
> data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>
>        <action application="socket" data="127.0.0.1:8084 async full"/>
>
>    </condition>
>
>  </extension>
>
>
>
> The above dialplan sample would invoke ivr-demo.php when 55522 is
> called as long as fs_ivrd is running. To start fs_ivrd:
>
> /usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8004
>
>
>
> It takes 2 arguments -h for hostname and -p for port.
>
>
>
> PHP Code
>
> #!/usr/bin/php -q
>
> <?php
>
>
>
>    // set a couple of things so we dont kill the system
>
>    ob_implicit_flush(true);
>
>    set_time_limit(30);
>
>
>
>
>
>    // Open stdin so we can read the AGI data in
>
>    $in = fopen("php://stdin", "r");
>
>    // Connect
>
>    echo "connect\n\n";
>
>    // Answer
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: answer\n\n";
>
>
>
>    // Play a prompt
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: playback\n";
>
>    echo "execute-app-arg:
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr- 
> welcome_to_freeswitch.wav\n\n";
>
>
>
>    // Wait
>
>    sleep(5);
>
>
>
>   // Hangup
>
>   echo "sendmsg\n";
>
>   echo "call-command: hangup\n\n";
>
>
>
> fclose($in);
>
>
>
> ?>
>
>
>
> ivrd will call this script for each call. All itdoes is answer the
> channel tell FreeSWITCH to play the ?welcome to freeswitch? prompt.
> Since the script is now controlling all call flow I needed to add a
> wait or it would send the hangup immediately before the prompt was
> played.
>
> Some improvements possible but that's 1 way to do it. It would be
> possible to do the socket directly in PHP but fs_ivrd is a nice option
> too.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at dialyourleads.com  Mon Aug  3 12:23:04 2009
From: mike at dialyourleads.com (Michael Frager)
Date: Mon, 3 Aug 2009 15:23:04 -0400
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
Message-ID: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>

Hello,

I'm in the process of moving my VOIP application from Asterisk to
FreeSWITCH.

I was wondering if it is possible to emulate the call announcement feature
that is available on Asterisk.

On Asterisk it looks like this, with the "A(...)" parameter:

Dial(SIP/15555551212|180|A(connecttone1))

Note that this announcement is only played for the called party, the calling
party does NOT hear the tone.

I'm guessing this can be done with FreeSWITCH.

Does anyone know how I might accomplish this?


Thanks in advance,

-Mike Fragre
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From david.nembrot at sogeti.com  Wed Aug  5 01:18:30 2009
From: david.nembrot at sogeti.com (David Nembrot)
Date: Wed,  5 Aug 2009 10:18:30 +0200
Subject: [Freeswitch-users] Qualify IM comm. across two distinct SIP domains
Message-ID: <20090805101830.bn9e3vzjms08oskg@mail.sogeti.com>

Hi everybody,

? ?I've just configured two Freeswitch servers (FS#1 and FS#2) to?
enable SIP communications between their two distinct SIP domains. The?
fact is that the IP telephony is up & running across these two?
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory?
way, my guess is that I'm missing something in the config files in?
order to enable IM services throughout the two domains.. Since they?
are in different networks, it seems reasonable for example to force?
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my?
question:

? ?How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

? ?or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.
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   Hi everybody,

   I've just configured two Freeswitch servers (FS#1 and FS#2) to  
enable SIP communications between their two distinct SIP domains. The  
fact is that the IP telephony is up & running across these two  
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory  
way, my guess is that I'm missing something in the config files in  
order to enable IM services throughout the two domains.. Since they  
are in different networks, it seems reasonable for example to force  
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my  
question:

   How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

   or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.


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From enno.egbert at googlemail.com  Wed Aug  5 01:52:20 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Wed, 5 Aug 2009 01:52:20 -0700 (PDT)
Subject: [Freeswitch-users]  TLS/SRTP with Innovaphone IP200
Message-ID: <24823167.post@talk.nabble.com>


Hello,

i have a problem using a innovaphone ip200 with freeswitch and tls/srtp. The
freeswitch certificate is in the trust list of the phone and it works with
tls for incomming calls. But outgoing calls were rejected to the mailbox.
The freeswitch configuration is ok, because it works with a snom 320.

Who can help me to confugure the IP200?

Thanks

NOx
-- 
View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From ovh at 5ca.com  Wed Aug  5 04:22:10 2009
From: ovh at 5ca.com (Otto)
Date: Wed, 5 Aug 2009 08:22:10 -0300
Subject: [Freeswitch-users] Looking for some FreeSWITCH job
Message-ID: <6B7F51E2C75FB1458C416CFEAD0E822F5D9E5B@ml.5CA.INT>

Diego,

 

You can contact me.

 

Br.

 

Otto van Haaren

www.5ca.com <http://www.5ca.com/> 

ovh at 5ca.com

 

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Diego Viola
Sent: dinsdag 4 augustus 2009 23:22
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Looking for some FreeSWITCH job

 

Hi,

I'm currently looking for some FS jobs, I really need one, I'm currently
unemployed and looking for some serious FreeSWITCH jobs.

Anyone?

P.S: I also do FS and web development with any language, PHP, Ruby, etc.
anything really.

Thanks,

Diego

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From david.nembrot at sogeti.com  Wed Aug  5 05:32:10 2009
From: david.nembrot at sogeti.com (David Nembrot)
Date: Wed,  5 Aug 2009 14:32:10 +0200
Subject: [Freeswitch-users] Qualify IM comm. across two distinct SIP domains
Message-ID: <20090805143210.um73qlhsgck80kko@mail.sogeti.com>

Hi everybody,

? ?I've just configured two Freeswitch servers (FS#1 and FS#2) to?
enable SIP communications between their two distinct SIP domains. The?
fact is that the IP telephony is up & running across these two?
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory?
way, my guess is that I'm missing something in the config files in?
order to enable IM services throughout the two domains.. Since they?
are in different networks, it seems reasonable for example to force?
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my?
question:

? ?How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

? ?or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.
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-------------- next part --------------
Hi everybody,

? ?I've just configured two Freeswitch servers (FS#1 and FS#2) to?
enable SIP communications between their two distinct SIP domains. The?
fact is that the IP telephony is up & running across these two?
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory?
way, my guess is that I'm missing something in the config files in?
order to enable IM services throughout the two domains.. Since they?
are in different networks, it seems reasonable for example to force?
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my?
question:

? ?How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

? ?or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.
-------------- next part --------------
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-------------- next part --------------


   Hi everybody,

   I've just configured two Freeswitch servers (FS#1 and FS#2) to  
enable SIP communications between their two distinct SIP domains. The  
fact is that the IP telephony is up & running across these two  
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory  
way, my guess is that I'm missing something in the config files in  
order to enable IM services throughout the two domains.. Since they  
are in different networks, it seems reasonable for example to force  
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my  
question:

   How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

   or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.


-------------- next part --------------
[Pi??ce jointe retir??e??: type d'origine de la pi??ce jointe: "text/html", nom: "Version_HTML_du_message"]

From dujinfang at gmail.com  Wed Aug  5 06:15:32 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Aug 2009 21:15:32 +0800
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
Message-ID: <EC77D164-BAAD-48FA-A3C4-6BFE05CC2E8F@gmail.com>

I think you can check the loopback endpoint or inline dialplan.

On Aug 4, 2009, at 3:23 AM, Michael Frager wrote:
> Hello,
>
> I'm in the process of moving my VOIP application from Asterisk to  
> FreeSWITCH.
>
> I was wondering if it is possible to emulate the call announcement  
> feature that is available on Asterisk.
>
> On Asterisk it looks like this, with the "A(...)" parameter:
>
> Dial(SIP/15555551212|180|A(connecttone1))
>
> Note that this announcement is only played for the called party, the  
> calling party does NOT hear the tone.
>
> I'm guessing this can be done with FreeSWITCH.
>
> Does anyone know how I might accomplish this?
>
>
> Thanks in advance,
>
> -Mike Fragre
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pjintheusa at gmail.com  Wed Aug  5 06:20:16 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 5 Aug 2009 09:20:16 -0400
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
Message-ID: <367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>

Mike,

I am not familiar with Asterisk so I am not 100% sure this is what you are
looking for, but check out
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

It may contain what you need

Phillip Jones


On Mon, Aug 3, 2009 at 3:23 PM, Michael Frager <mike at dialyourleads.com>wrote:

> Hello,
>
> I'm in the process of moving my VOIP application from Asterisk to
> FreeSWITCH.
>
> I was wondering if it is possible to emulate the call announcement feature
> that is available on Asterisk.
>
> On Asterisk it looks like this, with the "A(...)" parameter:
>
> Dial(SIP/15555551212|180|A(connecttone1))
>
> Note that this announcement is only played for the called party, the
> calling party does NOT hear the tone.
>
> I'm guessing this can be done with FreeSWITCH.
>
> Does anyone know how I might accomplish this?
>
>
> Thanks in advance,
>
> -Mike Fragre
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Aug  5 06:22:52 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 08:22:52 -0500
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <24823167.post@talk.nabble.com>
References: <24823167.post@talk.nabble.com>
Message-ID: <DF06C882-055F-488D-B860-46643AF9B22B@freeswitch.org>

Can you ship me a phone to test with?  That's usually the missing  
element when testing this stuff is I just can't afford to buy every  
phone to test with.

/b

On Aug 5, 2009, at 3:52 AM, NOx-WHV wrote:

>
> Hello,
>
> i have a problem using a innovaphone ip200 with freeswitch and tls/ 
> srtp. The
> freeswitch certificate is in the trust list of the phone and it  
> works with
> tls for incomming calls. But outgoing calls were rejected to the  
> mailbox.
> The freeswitch configuration is ok, because it works with a snom 320.
>
> Who can help me to confugure the IP200?
>
> Thanks
>
> NOx
> -- 



From brian at freeswitch.org  Wed Aug  5 06:23:48 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 08:23:48 -0500
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
	<367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>
Message-ID: <488F9396-69AF-4A50-943D-DD554F946741@freeswitch.org>

That is exactly what he's lookin for.

/b

On Aug 5, 2009, at 8:20 AM, Phillip Jones wrote:

> Mike,
>
> I am not familiar with Asterisk so I am not 100% sure this is what  
> you are looking for, but check out http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>
> It may contain what you need
>
> Phillip Jones

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From msc at freeswitch.org  Wed Aug  5 06:40:52 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 08:40:52 -0500
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <488F9396-69AF-4A50-943D-DD554F946741@freeswitch.org>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
	<367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>
	<488F9396-69AF-4A50-943D-DD554F946741@freeswitch.org>
Message-ID: <87f2f3b90908050640x5ec18d4ag417f38e31108b301@mail.gmail.com>

Could someone please add this to the FreeSWITCH wiki's Rosetta Stone page?
Thanks!-MC

On Wed, Aug 5, 2009 at 8:23 AM, Brian West <brian at freeswitch.org> wrote:

> That is exactly what he's lookin for.
> /b
>
> On Aug 5, 2009, at 8:20 AM, Phillip Jones wrote:
>
> Mike,
>
> I am not familiar with Asterisk so I am not 100% sure this is what you are
> looking for, but check out
> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>
> It may contain what you need
>
> Phillip Jones
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From max.bridgewater at gmail.com  Wed Aug  5 06:56:39 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 5 Aug 2009 09:56:39 -0400
Subject: [Freeswitch-users] How to distinguish between the different type of
	call rejections from Javascript?
Message-ID: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>

Hi,

Say i originate a call to a mobile phone and the call fails. There are many
possible reasons: congestion, user busy, call rejected by user, etc. Is
there a way i can get the failure code from Javascript?
Thanks,
Max.
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From dujinfang at gmail.com  Wed Aug  5 07:02:50 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Aug 2009 22:02:50 +0800
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
Message-ID: <D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>

variable_originate_disposition

On Aug 5, 2009, at 9:56 PM, Max Bridgewater wrote:
> Hi,
>
> Say i originate a call to a mobile phone and the call fails. There  
> are many possible reasons: congestion, user busy, call rejected by  
> user, etc. Is there a way i can get the failure code from Javascript?
> Thanks,
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jim at evolutiontel.net  Wed Aug  5 07:07:50 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Thu, 6 Aug 2009 00:07:50 +1000
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <24823167.post@talk.nabble.com>
References: <24823167.post@talk.nabble.com>
Message-ID: <ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>

Hi NOx,

Can you clarify the direction of the calls.  When you say outgoing do
you mean a call is terminating to the ip200?

I have been down a similar path while testing Eyebeam.  If the
terminating phone sets an option to only accept secure calls and FS
does not send Secure Descriptions in the INVITE, Eyebeam would respond
with 415 response code and the call would fail.  Depending on your
diaplan this could send your call to voicemail.

To fix it I added the following code to dialplan.
<action application="set" data="continue_on_fail=79"/>
<bridge blah blah blah>
<action application="set" data="bypass_media=false"/>
<action application="set" data="proxy_media=true"/>
<action application="set" data="ringback=$${uk-ring}"/>
<action application="pre_answer"/>
<action application="export" data="sip_secure_media=true"/>
<bridge blah blah blah>

The continue on fail captures the 415 response code forces the call to
continue to the next bridge while sip_secure_media forces the second
invite to include security descriptors.  The rest was required because
I did not want to proxy media if the call was not secure, obviously if
the call is secure on a point to point basis FS will have to proxy the
media and this was the only way I could find for it to work.

Hope this helps.

Regards,


On Wed, Aug 5, 2009 at 6:52 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>
> Hello,
>
> i have a problem using a innovaphone ip200 with freeswitch and tls/srtp. The
> freeswitch certificate is in the trust list of the phone and it works with
> tls for incomming calls. But outgoing calls were rejected to the mailbox.
> The freeswitch configuration is ok, because it works with a snom 320.
>
> Who can help me to confugure the IP200?
>
> Thanks
>
> NOx
> --
> View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From gregt at cgicommunications.com  Wed Aug  5 07:40:31 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 10:40:31 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
Message-ID: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>

Trying to make phpmod and it fails with this:

/usr/bin/ld: cannot find -laspell
collect2: ld returned 1 exit status
make[1]: *** [ESL.so] Error 1

I do have php-devel on this Centos 5.2 machine. Any ideas?
--
Greg
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From brian at freeswitch.org  Wed Aug  5 07:46:51 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 09:46:51 -0500
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
Message-ID: <2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>

install aspell-devel

also Please don't hijack threads... please click new message and start  
a new thread.

Thanks,
Brian

On Aug 5, 2009, at 9:40 AM, Greg Thoen wrote:

> Trying to make phpmod and it fails with this:
>
> /usr/bin/ld: cannot find -laspell
> collect2: ld returned 1 exit status
> make[1]: *** [ESL.so] Error 1
>
> I do have php-devel on this Centos 5.2 machine. Any ideas?
> --
> Greg

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From william.suffill at gmail.com  Wed Aug  5 07:47:38 2009
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 5 Aug 2009 10:47:38 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> 
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> 
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca> 
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com> 
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
Message-ID: <6b65470d0908050747r46da5b14q6ac3f955ad110e71@mail.gmail.com>

You don't have the dev library for aspell

yum install aspell-devel

should do the tick.

-- W


From msc at freeswitch.org  Wed Aug  5 07:54:45 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 5 Aug 2009 09:54:45 -0500
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
References: <24823167.post@talk.nabble.com>
	<ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
Message-ID: <95500CDE-C8A9-4F37-958D-43F27B5A6B3F@freeswitch.org>

Jim,

Just curious - could you document this use case on the wiki? Maybe you  
could create a page describing the setup and then link to it from the  
TLS page.

Thanks,
MC

Sent from my iPhone

On Aug 5, 2009, at 9:07 AM, Jim Burke <jim at evolutiontel.net> wrote:

> Hi NOx,
>
> Can you clarify the direction of the calls.  When you say outgoing do
> you mean a call is terminating to the ip200?
>
> I have been down a similar path while testing Eyebeam.  If the
> terminating phone sets an option to only accept secure calls and FS
> does not send Secure Descriptions in the INVITE, Eyebeam would respond
> with 415 response code and the call would fail.  Depending on your
> diaplan this could send your call to voicemail.
>
> To fix it I added the following code to dialplan.
> <action application="set" data="continue_on_fail=79"/>
> <bridge blah blah blah>
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="proxy_media=true"/>
> <action application="set" data="ringback=$${uk-ring}"/>
> <action application="pre_answer"/>
> <action application="export" data="sip_secure_media=true"/>
> <bridge blah blah blah>
>
> The continue on fail captures the 415 response code forces the call to
> continue to the next bridge while sip_secure_media forces the second
> invite to include security descriptors.  The rest was required because
> I did not want to proxy media if the call was not secure, obviously if
> the call is secure on a point to point basis FS will have to proxy the
> media and this was the only way I could find for it to work.
>
> Hope this helps.
>
> Regards,
>
>
> On Wed, Aug 5, 2009 at 6:52 PM, NOx-WHV<enno.egbert at googlemail.com>  
> wrote:
>>
>> Hello,
>>
>> i have a problem using a innovaphone ip200 with freeswitch and tls/ 
>> srtp. The
>> freeswitch certificate is in the trust list of the phone and it  
>> works with
>> tls for incomming calls. But outgoing calls were rejected to the  
>> mailbox.
>> The freeswitch configuration is ok, because it works with a snom 320.
>>
>> Who can help me to confugure the IP200?
>>
>> Thanks
>>
>> NOx
>> --
>> View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From helmut.kuper at ewetel.de  Wed Aug  5 07:56:30 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 05 Aug 2009 16:56:30 +0200
Subject: [Freeswitch-users] pocketsphinx
In-Reply-To: <87f2f3b90907311145p6c8d5907mc4545c710e9605fb@mail.gmail.com>
References: <4A5739DE.1080800@ewetel.de> <4A72EF33.4070404@ewetel.de>
	<87f2f3b90907311145p6c8d5907mc4545c710e9605fb@mail.gmail.com>
Message-ID: <4A799D9E.2020103@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

today I put my documentation on FS wiki describing the steps to get a
8kHz sample rate acoustic model basing on voxforge's data for german
language. It's not complete, yet.

You can found it here: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx

regards
Helmut


On 31.07.2009 20:45, Michael Collins wrote:
> Helmut,
> 
> Your hard work is appreciated. Like Brian said, we'd all be interested
> in knowing more. Please feel free to put this on the wiki or see me off
> list and we'll discuss further how to document it for the good of the FS
> community.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFKeZ2e4tZeNddg3dwRAlwuAKCXK6b/f3J7tRmcev0/EPAUFGZBbgCfXMQW
B8MAREKeR82dTFnYyFeutig=
=1/Jr
-----END PGP SIGNATURE-----


From woodydickson at gmail.com  Wed Aug  5 08:05:20 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 5 Aug 2009 23:05:20 +0800
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
Message-ID: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>

Hi,

I want to implement a module where freeSWITCH would try to bridge to an
extension and if the bridging operation fails, my module can use the hangup
code to determine the next cause of action.

With switch_caller_extension_add_application(session, extension, "bridge",
"sofia/gateway/mygw/1232323);, if there is an error ( 503 received for
instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the
module's APP) and go on to the next action.  Is there anyway to control it
so that freeSWITCH would remain to be within the module's APP funtion and
continue executing the code after switch_call_extension_add_application,
when let's say a 4XX or 5XX or CANCEL ( from originator) is received?

I have tried it and found that if the bridging is successful, freeSWITCH
would continue executing the code after
switch_caller_extension_add_application, but if an error is received, then
it would just move on to the next action.

Does anyone know how to deal with this problem?

Woody
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From gregt at cgicommunications.com  Wed Aug  5 08:21:41 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 11:21:41 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
	<2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>
Message-ID: <BFC0A2A3-F0D5-40DC-8E1B-AA2FB2FFB372@cgicommunications.com>

Oops, what thread did I hijack? I just sent a new email message to freeswitch-users at lists.freeswitch.org
What should I have done?
--
Greg


On Aug 5, 2009, at 10:46 AM, Brian West wrote:

> install aspell-devel
>
> also Please don't hijack threads... please click new message and  
> start a new thread.
>
> Thanks,
> Brian
>
> On Aug 5, 2009, at 9:40 AM, Greg Thoen wrote:
>
>> Trying to make phpmod and it fails with this:
>>
>> /usr/bin/ld: cannot find -laspell
>> collect2: ld returned 1 exit status
>> make[1]: *** [ESL.so] Error 1
>>
>> I do have php-devel on this Centos 5.2 machine. Any ideas?
>> --
>> Greg
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Wed Aug  5 08:39:31 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 10:39:31 -0500
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <BFC0A2A3-F0D5-40DC-8E1B-AA2FB2FFB372@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
	<2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>
	<BFC0A2A3-F0D5-40DC-8E1B-AA2FB2FFB372@cgicommunications.com>
Message-ID: <44BA0B61-3971-479B-816C-5501035DB28D@freeswitch.org>

Re: [Freeswitch-users] ESL Wrapper



It happens when you click reply... change the subject and the body...  
thats how you hijack a thread.  :)  Happens to the best of us.



/b





On Aug 5, 2009, at 10:21 AM, Greg Thoen wrote:

> Oops, what thread did I hijack? I just sent a new email message to freeswitch-users at lists.freeswitch.org
> What should I have done?
> --
> Greg
>
>
> On Aug 5, 2009, at 10:46 AM, Brian West wrote:
>
>> install aspell-devel
>>
>> also Please don't hijack threads... please click new message and  
>> start a new thread.
>>
>> Thanks,
>> Brian
>>
>> On Aug 5, 2009, at 9:40 AM, Greg Thoen wrote:
>>
>>> Trying to make phpmod and it fails with this:
>>>
>>> /usr/bin/ld: cannot find -laspell
>>> collect2: ld returned 1 exit status
>>> make[1]: *** [ESL.so] Error 1
>>>
>>> I do have php-devel on this Centos 5.2 machine. Any ideas?
>>> --
>>> Greg
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jmesquita at gmail.com  Wed Aug  5 09:15:30 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Aug 2009 13:15:30 -0300
Subject: [Freeswitch-users] How to change the contact when fs sending
	REGISTER??
In-Reply-To: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
References: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
Message-ID: <5a8712120908050915k2f414a37vd651f9323114cdbf@mail.gmail.com>

Add the following line to the gw definition:

<param name="extension-in-contact" value="true"/>

jmesquita

On Wed, Aug 5, 2009 at 7:26 AM, Brad Tuan <brad.tuan at gmail.com> wrote:

> As title ,I know how to do when sending INVITE
>
> but how to do it when fs sending REGISTER??
>
> For example , when gateway registering , the contact is
> gw+abcd at XXX.XXX.XXX.XXX ,
>
> how to change it to *abcd at XXX.XXX.XXX.XXX??*
> **
> *Please help*
> **
> **
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at gmail.com  Wed Aug  5 09:23:26 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Aug 2009 13:23:26 -0300
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
Message-ID: <5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>

My guess is that you will receive a message here:

switch_status_t channel_receive_message(switch_core_session_t *session,
switch_core_session_message_t *msg)

The problem here is that you don't have the exact SIP code but there is a
clear relationship between the codes and the messages you receive on the
channel, so I am guessing that is all the same.

Hope this helps.

jmesquita

On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I want to implement a module where freeSWITCH would try to bridge to an
> extension and if the bridging operation fails, my module can use the hangup
> code to determine the next cause of action.
>
> With switch_caller_extension_add_application(session, extension, "bridge",
> "sofia/gateway/mygw/1232323);, if there is an error ( 503 received for
> instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the
> module's APP) and go on to the next action.  Is there anyway to control it
> so that freeSWITCH would remain to be within the module's APP funtion and
> continue executing the code after switch_call_extension_add_application,
> when let's say a 4XX or 5XX or CANCEL ( from originator) is received?
>
> I have tried it and found that if the bridging is successful, freeSWITCH
> would continue executing the code after
> switch_caller_extension_add_application, but if an error is received, then
> it would just move on to the next action.
>
> Does anyone know how to deal with this problem?
>
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Wed Aug  5 09:34:50 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 5 Aug 2009 11:34:50 -0500
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
Message-ID: <E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>

The hangup cause will be in the originate_disposition channel variable  
on the A-leg.

sip_term_status will contain the sip code and  
proto_specific_hangup_cause will contain sip:<code>.


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:

> My guess is that you will receive a message here:
>
> switch_status_t channel_receive_message(switch_core_session_t  
> *session, switch_core_session_message_t *msg)
>
> The problem here is that you don't have the exact SIP code but there  
> is a clear relationship between the codes and the messages you  
> receive on the channel, so I am guessing that is all the same.
>
> Hope this helps.
>
> jmesquita
>
> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson  
> <woodydickson at gmail.com> wrote:
> Hi,
>
> I want to implement a module where freeSWITCH would try to bridge to  
> an extension and if the bridging operation fails, my module can use  
> the hangup code to determine the next cause of action.
>
> With switch_caller_extension_add_application(session, extension,  
> "bridge", "sofia/gateway/mygw/1232323);, if there is an error ( 503  
> received for instance ) in the outgoing INVITE, freeSWITCH would  
> leave my module ( or the module's APP) and go on to the next  
> action.  Is there anyway to control it so that freeSWITCH would  
> remain to be within the module's APP funtion and continue executing  
> the code after switch_call_extension_add_application, when let's say  
> a 4XX or 5XX or CANCEL ( from originator) is received?
>
> I have tried it and found that if the bridging is successful,  
> freeSWITCH would continue executing the code after  
> switch_caller_extension_add_application, but if an error is  
> received, then it would just move on to the next action.
>
> Does anyone know how to deal with this problem?
>
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mayamatakeshi at gmail.com  Wed Aug  5 09:35:38 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 6 Aug 2009 01:35:38 +0900
Subject: [Freeswitch-users] Monitoring On-Hold/Off-Hold
Message-ID: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>

Hello,
I'm using mod_event_socket to monitor FS.
I'm using "events plain ALL' and I get lots of channel events. But
curiously, when some channel puts the call on-hold/off-hold, I don't
get any notification. Is it possible to get these events? Am I missing
some setting?

regards,
takeshi


From gregt at cgicommunications.com  Wed Aug  5 10:31:23 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 13:31:23 -0400
Subject: [Freeswitch-users] Best practice for inbound calls with scripting
Message-ID: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>

Hi. I am setting up a large inbound only system with multiple DIDs  
coming in; each call is processed fairly intensively with db lookups,  
wav files played, pocketsphinx is used, wav files recorded, etc.

Before I get too far down one path, I was wondering if anyone had any  
insight into the best, most scaleable way to do this out of the  
several methods I can do:

1	Call comes in
	dialplan calls specific javascript based on DID
	javascript uses ODBC to pull info from local mysql db
	call is handled in javascript

2	Call comes in
	dialplan calls specific javascript based on DID
	javascript uses CURL to get info from local mysql db
	call is handled in javascript

3	Call comes in
	php socket is listening for call
	php script runs, pulling info from mysql db
	call is handled in php using esl.php

4	Call comes in
	dialplan calls specific lua script  based on DID
	lua uses luasql.mysql to get info from local mysql db
	call is handled in lua using lua api

5	Call comes in
	xml_curl is used for dynamic dialplan
	js called, continues like #1

I know that they will all do essentially the same thing. But once I go  
down the path, I don't want to find out that the way I chose chokes  
with 50 simultaneous inbound calls on different DIDs. Any comments  
would be appreciated. Thanks.
--
Greg


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From jmesquita at gmail.com  Wed Aug  5 10:55:37 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Aug 2009 14:55:37 -0300
Subject: [Freeswitch-users] Monitoring On-Hold/Off-Hold
In-Reply-To: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>
References: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>
Message-ID: <5a8712120908051055k95f01d4x61ef7b7dff80298a@mail.gmail.com>

I only see one way out of this. If you manage presence, an event like the
following is sent:

Event-Name: PRESENCE_IN
Core-UUID: 189b12c0-7fb0-11de-b0bc-37eec03ad00f
FreeSWITCH-Hostname: cl-t146-421cl
FreeSWITCH-IPv4: XXXXXX
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-05%2013%3A42%3A24
Event-Date-GMT: Wed,%2005%20Aug%202009%2017%3A42%3A24%20GMT
Event-Date-Timestamp: 1249494144628132
Event-Calling-File: switch_channel.c
Event-Calling-Function: switch_channel_presence
Event-Calling-Line-Number: 472
Channel-State: CS_HIBERNATE
Channel-State-Number: 8
Channel-Name: XXXXX
Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
Call-Direction: inbound
Presence-Call-Direction: inbound
Answer-State: answered
Caller-Username: 1000
Caller-Dialplan: XML
Caller-Caller-ID-Name: Mesquita
Caller-Caller-ID-Number: 1000
Caller-Network-Addr: XXXXX
Caller-Destination-Number: 1005
Caller-Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
Caller-Source: mod_sofia
Caller-Context: XXXXX
Caller-Channel-Name: XXXXX
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249494132128119
Caller-Channel-Created-Time: 1249494132128119
Caller-Channel-Answered-Time: 1249494139500129
Caller-Channel-Progress-Time: 1249494132368119
Caller-Channel-Progress-Media-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Leg-Username: 1000
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: Joao%20Mesquita
Other-Leg-Caller-ID-Number: 1000
Other-Leg-Network-Addr: 190.2.41.65
Other-Leg-Destination-Number:
sip%3A1005%40192.168.0.106%3A4559%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1005%2540190.2.41.65%253A4559
Other-Leg-Unique-ID: 4e7622ac-81e7-11de-b0bc-37eec03ad00f
Other-Leg-Source: mod_sofia
Other-Leg-Context: XXXXX
Other-Leg-Channel-Name: XXXXXX
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false
proto: src/switch_channel.c
login: src/switch_channel.c
from: XXXXXX
rpid: unknown
status: hold
event_type: presence
alt_event_type: dialog
event_count: 3

Content-Length: 543
Content-Type: text/event-plain

Other than that, I think it can be patched. I will take a look at it.

Guys, should this be patched on the state machine itself or on the mod_sofia
channel_receive_message?

jmesquita


On Wed, Aug 5, 2009 at 1:35 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Hello,
> I'm using mod_event_socket to monitor FS.
> I'm using "events plain ALL' and I get lots of channel events. But
> curiously, when some channel puts the call on-hold/off-hold, I don't
> get any notification. Is it possible to get these events? Am I missing
> some setting?
>
> regards,
> takeshi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dule.maillist at gmail.com  Wed Aug  5 10:59:38 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Wed, 5 Aug 2009 13:59:38 -0400
Subject: [Freeswitch-users] Best practice for inbound calls with
	scripting
In-Reply-To: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
References: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
Message-ID: <914fc92a0908051059na35f58ar3f102ef4d80e0027@mail.gmail.com>

One common recommendation is to use lua over js, since it's lighter-weight,
using less resources.
Dan

On Wed, Aug 5, 2009 at 1:31 PM, Greg Thoen <gregt at cgicommunications.com>wrote:

> Hi. I am setting up a large inbound only system with multiple DIDs coming
> in; each call is processed fairly intensively with db lookups, wav files
> played, pocketsphinx is used, wav files recorded, etc.
> Before I get too far down one path, I was wondering if anyone had any
> insight into the best, most scaleable way to do this out of the several
> methods I can do:
> 1 Call comes in
> dialplan calls specific javascript based on DID
> javascript uses ODBC to pull info from local mysql db
> call is handled in javascript
>
> 2 Call comes in
> dialplan calls specific javascript based on DID
> javascript uses CURL to get info from local mysql db
> call is handled in javascript
>
> 3 Call comes in
> php socket is listening for call
> php script runs, pulling info from mysql db
> call is handled in php using esl.php
>
> 4 Call comes in
> dialplan calls specific lua script  based on DID
> lua uses luasql.mysql to get info from local mysql db
> call is handled in lua using lua api
>
> 5 Call comes in
> xml_curl is used for dynamic dialplan
> js called, continues like #1
>
> I know that they will all do
> essentially the same thing. But once I go down the path, I don't want to find out that the way I chose chokes with 50 simultaneous inbound calls on different DIDs. Any comments would be appreciated. Thanks.
>
> --
>
> Greg
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From technical at ttnc.co.uk  Wed Aug  5 11:20:46 2009
From: technical at ttnc.co.uk (TTNC - Adnan Barakat)
Date: Wed, 05 Aug 2009 19:20:46 +0100
Subject: [Freeswitch-users] Best practice for inbound calls with
	scripting
In-Reply-To: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
References: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
Message-ID: <4A79CD7E.6010806@ttnc.co.uk>

Greg Thoen wrote:
> 4 Call comes in
> dialplan calls specific lua script  based on DID
> lua uses luasql.mysql to get info from local mysql db
> call is handled in lua using lua api
This is the option we chose in our set-up (although mysql is on a remote 
server), currently we have over 25k DIDs running on FreeSWITCH via this 
set-up and everything is working wonderfully.

We were using option 5 previously, but found 4 to be much better in general.

Hope this helps.


Adnan


From edpimentl at gmail.com  Wed Aug  5 11:48:16 2009
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 5 Aug 2009 14:48:16 -0400
Subject: [Freeswitch-users] Best practice for inbound calls with
	scripting
In-Reply-To: <914fc92a0908051059na35f58ar3f102ef4d80e0027@mail.gmail.com>
References: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
	<914fc92a0908051059na35f58ar3f102ef4d80e0027@mail.gmail.com>
Message-ID: <9dc4a1670908051148g38958616j57c2a7eebf9f607f@mail.gmail.com>

Here are other scenarios to add

Call request comes in via SMS
Call request comes in via XMPP
Call request comes in via EMAIL
Call request comes in via TWITTER/FACEBOOK/WAVE/

Best regards,
-E
Gpro.ws
edpimentl [SKype ]



On Wed, Aug 5, 2009 at 1:59 PM, Dan Le <dule.maillist at gmail.com> wrote:

> One common recommendation is to use lua over js, since it's lighter-weight,
> using less resources.
> Dan
>
> On Wed, Aug 5, 2009 at 1:31 PM, Greg Thoen <gregt at cgicommunications.com>wrote:
>
>> Hi. I am setting up a large inbound only system with multiple DIDs coming
>> in; each call is processed fairly intensively with db lookups, wav files
>> played, pocketsphinx is used, wav files recorded, etc.
>> Before I get too far down one path, I was wondering if anyone had any
>> insight into the best, most scaleable way to do this out of the several
>> methods I can do:
>> 1 Call comes in
>>  dialplan calls specific javascript based on DID
>> javascript uses ODBC to pull info from local mysql db
>> call is handled in javascript
>>
>> 2 Call comes in
>> dialplan calls specific javascript based on DID
>> javascript uses CURL to get info from local mysql db
>>  call is handled in javascript
>>
>> 3 Call comes in
>> php socket is listening for call
>>  php script runs, pulling info from mysql db
>> call is handled in php using esl.php
>>
>> 4 Call comes in
>>  dialplan calls specific lua script  based on DID
>> lua uses luasql.mysql to get info from local mysql db
>> call is handled in lua using lua api
>>
>> 5 Call comes in
>> xml_curl is used for dynamic dialplan
>> js called, continues like #1
>>
>> I know that they will all do
>> essentially the same thing. But once I go down the path, I don't want to find out that the way I chose chokes with 50 simultaneous inbound calls on different DIDs. Any comments would be appreciated. Thanks.
>>
>> --
>>
>> Greg
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kjoseph.us at gmail.com  Wed Aug  5 13:24:50 2009
From: kjoseph.us at gmail.com (Joseph Khoury)
Date: Wed, 5 Aug 2009 13:24:50 -0700
Subject: [Freeswitch-users] Looking for some FreeSWITCH job
In-Reply-To: <6B7F51E2C75FB1458C416CFEAD0E822F5D9E5B@ml.5CA.INT>
References: <6B7F51E2C75FB1458C416CFEAD0E822F5D9E5B@ml.5CA.INT>
Message-ID: <7d1481c30908051324j5b806cb6y527162037470fead@mail.gmail.com>

Diego,
Contact me also.

Joseph

www.alosmart.com
(myname) at alosmart.com

On Wed, Aug 5, 2009 at 4:22 AM, Otto <ovh at 5ca.com> wrote:

>  Diego,
>
>
>
> You can contact me.
>
>
>
> Br.
>
>
>
> Otto van Haaren
>
> www.5ca.com
>
> ovh at 5ca.com
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* dinsdag 4 augustus 2009 23:22
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Looking for some FreeSWITCH job
>
>
>
> Hi,
>
> I'm currently looking for some FS jobs, I really need one, I'm currently
> unemployed and looking for some serious FreeSWITCH jobs.
>
> Anyone?
>
> P.S: I also do FS and web development with any language, PHP, Ruby, etc.
> anything really.
>
> Thanks,
>
> Diego
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tomabroad at gmail.com  Wed Aug  5 11:46:37 2009
From: tomabroad at gmail.com (tom)
Date: Wed, 5 Aug 2009 14:46:37 -0400
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
Message-ID: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>

hi just installed freeswitch via svn.
- bootstrap
- configure
- make install
- ./freeswitch

gives me:
acerdebian:/usr/local/freeswitch/bin# ./freeswitch
Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch
-waste.
auto-adjusting stack size for optimal performance....
2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
Engine.
2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event dispatch
thread 0
2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)
Cannot Initialize [Cannot Open log directory or XML Root!]


bump - help

thx
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From mattdfong at gmail.com  Wed Aug  5 13:52:13 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 5 Aug 2009 13:52:13 -0700
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
In-Reply-To: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
References: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
Message-ID: <4256bf830908051352t26753d57nbeb6ba27f2a5f77a@mail.gmail.com>

Does the file exist at /usr/local/freeswitch/conf/freeswitch.xml? does the
user you are executing freeswitch as have permission to read the file?
--matt
hello hunter - hosted predictive dialer & voice broadcasting
http://www.hellohunter.com

On Wed, Aug 5, 2009 at 11:46 AM, tom <tomabroad at gmail.com> wrote:

> hi just installed freeswitch via svn.
> - bootstrap
> - configure
> - make install
> - ./freeswitch
>
> gives me:
> acerdebian:/usr/local/freeswitch/bin# ./freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
> ./freeswitch -waste.
> auto-adjusting stack size for optimal performance....
> 2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
> Engine.
> 2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event dispatch
> thread 0
> 2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt
> open /usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)

Cannot Initialize [Cannot Open log directory or XML Root!]
>
>
> bump - help
>
> thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sprice at gmail.com  Wed Aug  5 13:53:16 2009
From: sprice at gmail.com (SP)
Date: Wed, 5 Aug 2009 15:53:16 -0500
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
In-Reply-To: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
References: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
Message-ID: <7e2ac3270908051353m33c3f158r4bd7ec2d415f7a7a@mail.gmail.com>

2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/freeswitch.xml *(No such file or directory)*

On Wed, Aug 5, 2009 at 13:46, tom<tomabroad at gmail.com> wrote:
> hi just installed freeswitch via svn.
> - bootstrap
> - configure
> - make install
> - ./freeswitch
>
> gives me:
> acerdebian:/usr/local/freeswitch/bin# ./freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
./freeswitch
> -waste.
> auto-adjusting stack size for optimal performance....
> 2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
> Engine.
> 2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event
dispatch
> thread 0
> 2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
> /usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)
> Cannot Initialize [Cannot Open log directory or XML Root!]
>
>
> bump - help
>
> thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Shannon
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From raffaele.p.guidi at gmail.com  Wed Aug  5 13:54:46 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Wed, 5 Aug 2009 22:54:46 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
Message-ID: <b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>

interesting! what values can contain "variable_originate_disposition"? And
can I set them manually in a script to reject a call simulating user busy or
call rejected? A lua example?
Thanks,
   Raffaele

On Wed, Aug 5, 2009 at 16:02, Seven Du <dujinfang at gmail.com> wrote:

> variable_originate_disposition
>
> On Aug 5, 2009, at 9:56 PM, Max Bridgewater wrote:
> > Hi,
> >
> > Say i originate a call to a mobile phone and the call fails. There
> > are many possible reasons: congestion, user busy, call rejected by
> > user, etc. Is there a way i can get the failure code from Javascript?
> > Thanks,
> > Max.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Aug  5 13:59:07 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 15:59:07 -0500
Subject: [Freeswitch-users] How to change the contact when fs sending
	REGISTER??
In-Reply-To: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
References: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
Message-ID: <978FA06A-8591-44DF-BC08-5CAC2668F6ED@freeswitch.org>

My first question is why would you have to change it?  :P

/b

On Aug 5, 2009, at 5:26 AM, Brad Tuan wrote:

> As title ,I know how to do when sending INVITE
>
> but how to do it when fs sending REGISTER??
>
> For example , when gateway registering , the contact is gw+abcd at XXX.XXX.XXX.XXX 
>  ,
>
> how to change it to abcd at XXX.XXX.XXX.XXX??
>
> Please help
>

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From gmaruzz at celliax.org  Wed Aug  5 14:00:17 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 5 Aug 2009 23:00:17 +0200
Subject: [Freeswitch-users] freepbx for freeswitch
Message-ID: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>

Yay!

http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future

Darren Schreiber has made the announcement and is doinng a
presentation of FreePBX V3 right now at www.cluecon.com.


From tomabroad at gmail.com  Wed Aug  5 14:09:37 2009
From: tomabroad at gmail.com (tom)
Date: Wed, 5 Aug 2009 17:09:37 -0400
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
In-Reply-To: <7e2ac3270908051353m33c3f158r4bd7ec2d415f7a7a@mail.gmail.com>
References: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
	<7e2ac3270908051353m33c3f158r4bd7ec2d415f7a7a@mail.gmail.com>
Message-ID: <6f7c60c40908051409h57277282m6f6e94af47b6d75d@mail.gmail.com>

solved via "make samples"

thx tom
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From nik.middleton at noblesolutions.co.uk  Wed Aug  5 14:24:47 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Wed, 5 Aug 2009 22:24:47 +0100
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>

I'd heard rumours that this was going to happen and it's great news and
good news for FS as well.  With a user friendly front end, FS is sure to
fly.  I have no doubt that this will be the first of many.

Regards,

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Giovanni Maruzzelli
Sent: 05 August 2009 22:00
To: freeswitch-users at lists.freeswitch.org;
freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-users] freepbx for freeswitch

Yay!

http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
ure

Darren Schreiber has made the announcement and is doinng a
presentation of FreePBX V3 right now at www.cluecon.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From msc at freeswitch.org  Wed Aug  5 14:35:30 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 16:35:30 -0500
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
Message-ID: <87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>

Of course, don't forget this:
http://www.cudatel.com

For those who want a commercial solution built upon FreeSWITCH: You've got
it!
-MC

On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

> I'd heard rumours that this was going to happen and it's great news and
> good news for FS as well.  With a user friendly front end, FS is sure to
> fly.  I have no doubt that this will be the first of many.
>
> Regards,
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Giovanni Maruzzelli
> Sent: 05 August 2009 22:00
> To: freeswitch-users at lists.freeswitch.org;
> freeswitch-dev at lists.freeswitch.org
> Subject: [Freeswitch-users] freepbx for freeswitch
>
> Yay!
>
> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
> ure
>
> Darren Schreiber has made the announcement and is doinng a
> presentation of FreePBX V3 right now at www.cluecon.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Wed Aug  5 14:49:49 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 5 Aug 2009 17:49:49 -0400
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
	<87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
Message-ID: <86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>

Yay for FreePBX =D

Does Cudatel uses FreeSWITCH as the engine?

On Wed, Aug 5, 2009 at 5:35 PM, Michael Collins<msc at freeswitch.org> wrote:
> Of course, don't forget this:
> http://www.cudatel.com
>
> For those who want a commercial solution built upon FreeSWITCH: You've got
> it!
> -MC
>
> On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton
> <nik.middleton at noblesolutions.co.uk> wrote:
>>
>> I'd heard rumours that this was going to happen and it's great news and
>> good news for FS as well. ?With a user friendly front end, FS is sure to
>> fly. ?I have no doubt that this will be the first of many.
>>
>> Regards,
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Giovanni Maruzzelli
>> Sent: 05 August 2009 22:00
>> To: freeswitch-users at lists.freeswitch.org;
>> freeswitch-dev at lists.freeswitch.org
>> Subject: [Freeswitch-users] freepbx for freeswitch
>>
>> Yay!
>>
>> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
>> ure
>>
>> Darren Schreiber has made the announcement and is doinng a
>> presentation of FreePBX V3 right now at www.cluecon.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sprice at gmail.com  Wed Aug  5 14:59:31 2009
From: sprice at gmail.com (SP)
Date: Wed, 5 Aug 2009 16:59:31 -0500
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
	<87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
	<86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
Message-ID: <7e2ac3270908051459l16db1152k8eb6734add1957a3@mail.gmail.com>

would it be pimped if it didn't?

On Wed, Aug 5, 2009 at 16:49, Diego Viola<diego.viola at gmail.com> wrote:
> Yay for FreePBX =D
>
> Does Cudatel uses FreeSWITCH as the engine?
>
> On Wed, Aug 5, 2009 at 5:35 PM, Michael Collins<msc at freeswitch.org> wrote:
>> Of course, don't forget this:
>> http://www.cudatel.com
>>
>> For those who want a commercial solution built upon FreeSWITCH: You've got
>> it!
>> -MC
>>
>> On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton
>> <nik.middleton at noblesolutions.co.uk> wrote:
>>>
>>> I'd heard rumours that this was going to happen and it's great news and
>>> good news for FS as well. ?With a user friendly front end, FS is sure to
>>> fly. ?I have no doubt that this will be the first of many.
>>>
>>> Regards,
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Giovanni Maruzzelli
>>> Sent: 05 August 2009 22:00
>>> To: freeswitch-users at lists.freeswitch.org;
>>> freeswitch-dev at lists.freeswitch.org
>>> Subject: [Freeswitch-users] freepbx for freeswitch
>>>
>>> Yay!
>>>
>>> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
>>> ure
>>>
>>> Darren Schreiber has made the announcement and is doinng a
>>> presentation of FreePBX V3 right now at www.cluecon.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon


From msc at freeswitch.org  Wed Aug  5 15:00:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:00:06 -0500
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
	<87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
	<86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
Message-ID: <87f2f3b90908051500u59e5f6ack4402818e226df7a7@mail.gmail.com>

On Wed, Aug 5, 2009 at 4:49 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Yay for FreePBX =D
>
> Does Cudatel uses FreeSWITCH as the engine?


Yes it does!
-MC
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From msc at freeswitch.org  Wed Aug  5 15:02:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:02:28 -0500
Subject: [Freeswitch-users] pocketsphinx
In-Reply-To: <4A799D9E.2020103@ewetel.de>
References: <4A5739DE.1080800@ewetel.de> <4A72EF33.4070404@ewetel.de>
	<87f2f3b90907311145p6c8d5907mc4545c710e9605fb@mail.gmail.com>
	<4A799D9E.2020103@ewetel.de>
Message-ID: <87f2f3b90908051502o3943cbcdy35c91cd11174d743@mail.gmail.com>

On Wed, Aug 5, 2009 at 9:56 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Michael,
>
> today I put my documentation on FS wiki describing the steps to get a
> 8kHz sample rate acoustic model basing on voxforge's data for german
> language. It's not complete, yet.
>
> You can found it here: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx
>
> regards
> Helmut
>

Thanks!
I'd like to ask the community members who are interested in ASR and
PocketSphinx to please review Helmut's page and add to it as well as offer
feedback.
-MC
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From msc at freeswitch.org  Wed Aug  5 15:04:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:04:54 -0500
Subject: [Freeswitch-users] Fwd: Need Help In IVR
In-Reply-To: <7aa29e790908032247h2a93e838m1ac5fd3f8a3946c7@mail.gmail.com>
References: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>
	<7aa29e790908032247h2a93e838m1ac5fd3f8a3946c7@mail.gmail.com>
Message-ID: <87f2f3b90908051504i70dd6e6ag7bc5fe3fa0200dd2@mail.gmail.com>

Could you give us an update on what you have so far? How about you put your
dialplan and perl script into a pastebin so that we can get a good frame of
reference?
Thanks,
MC

On Tue, Aug 4, 2009 at 12:47 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Can you please help me?
>
>
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From msc at freeswitch.org  Wed Aug  5 15:16:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:16:41 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
Message-ID: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>

On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> interesting! what values can contain "variable_originate_disposition"? And
> can I set them manually in a script to reject a call simulating user busy or
> call rejected? A lua example?
> Thanks,
>    Raffaele
>

Start here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup

And note the link to the hangup causes. As far as Lua, I'm not sure there's
a good reason to do it there. Could you give us pseudo code example of what
you're thinking of doing?
-MC
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From raffaele.p.guidi at gmail.com  Wed Aug  5 16:19:10 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 6 Aug 2009 01:19:10 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
Message-ID: <b0a8792f0908051619s66caac7na886720273396bdd@mail.gmail.com>

Actually I was reading that page, right now. I wrote a small lua script that
simulates a call with random wait time before answering, randomly not
answering at all and saying things for a random times once answered. This
would be useful for testing purposes simulating load, letting call center
operators try scripts against "fake" numbers with a "realistic" behaviour
and eventually to test and debug an automated dialer.
The script is almost ready (can contribute it should you find it useful), I
used it today to simulate load on my windows laptop with 50 concurrent calls
and peaks of 20/30 simultaneous calls connected (cpu was below 3%). I only
miss some use cases such as some CALL_REJECTED, USER_BUSY, NO_ANSWER

On Thu, Aug 6, 2009 at 00:16, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> interesting! what values can contain "variable_originate_disposition"? And
>> can I set them manually in a script to reject a call simulating user busy or
>> call rejected? A lua example?
>> Thanks,
>>    Raffaele
>>
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure there's
> a good reason to do it there. Could you give us pseudo code example of what
> you're thinking of doing?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Aug  5 16:23:22 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 18:23:22 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
Message-ID: <121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>

The bigger problem is some end points won't hang up the call in a  
consistent manner... some phones say user_rejected or user_busy when  
you reject the call with the reject button.

/b

On Aug 5, 2009, at 5:16 PM, Michael Collins wrote:

>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <raffaele.p.guidi at gmail.com 
> > wrote:
> interesting! what values can contain  
> "variable_originate_disposition"? And can I set them manually in a  
> script to reject a call simulating user busy or call rejected? A lua  
> example?
>
> Thanks,
>    Raffaele
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure  
> there's a good reason to do it there. Could you give us pseudo code  
> example of what you're thinking of doing?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nicolas at medularis.com  Wed Aug  5 16:41:47 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 5 Aug 2009 19:41:47 -0400
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
Message-ID: <1b46b4e80908051641y170b8d07g98ac29f69250f2ce@mail.gmail.com>

I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:

2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL

But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
NORMAL_CLEARING. And the variable_originate_disposition has a value of
"failure". Where can I get the detail of knowing the call/bridge failed
because of 'ORIGINATOR_CANCEL' as reported through the console?

Thanks!

Nicolas


 the value of variable_originate_disposition at the events level and when I
have an origination failure due to 'ORIGINATOR_CANCEL

On Wed, Aug 5, 2009 at 6:16 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> interesting! what values can contain "variable_originate_disposition"? And
>> can I set them manually in a script to reject a call simulating user busy or
>> call rejected? A lua example?
>> Thanks,
>>    Raffaele
>>
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure there's
> a good reason to do it there. Could you give us pseudo code example of what
> you're thinking of doing?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From raffaele.p.guidi at gmail.com  Wed Aug  5 16:44:11 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 6 Aug 2009 01:44:11 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
Message-ID: <b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>

Well, I would randomly insert all of those cases to make it more
realistic... only thing I cannot manage to issue USER_BUSY from lua (and
neither from the dialplan, actually).

<anti-action application="respond" data="407" /> (407 or 486 or whatever...)


doesn't behave as I expected and neither

<action application="hangup" data="407" /> (407 or 486 or USER_BUSY or
whatever...)


and I cannot find a a session:reject(<hangupcause>) method in lua.

Can you give me a hint?

On Thu, Aug 6, 2009 at 01:23, Brian West <brian at freeswitch.org> wrote:

> The bigger problem is some end points won't hang up the call in a
> consistent manner... some phones say user_rejected or user_busy when you
> reject the call with the reject button.
> /b
>
> On Aug 5, 2009, at 5:16 PM, Michael Collins wrote:
>
>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> interesting! what values can contain "variable_originate_disposition"? And
>> can I set them manually in a script to reject a call simulating user busy or
>> call rejected? A lua example?
>> Thanks,
>>    Raffaele
>>
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure there's
> a good reason to do it there. Could you give us pseudo code example of what
> you're thinking of doing?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From woodydickson at gmail.com  Wed Aug  5 17:20:51 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 6 Aug 2009 08:20:51 +0800
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
Message-ID: <a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>

Hi,

The problem is that I need freeswitch to continue executing the code after
switch_status_t channel_receive_message even when it gets error SIP code
from the destination.  Is that possible?

I know if I set up another action after my module in the dialplan.xml, I can
catch that.

But I would like the code to execute within the route that I have.  Is that
doable?

Woody

On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> The hangup cause will be in the originate_disposition channel
> variable on the A-leg.
> sip_term_status will contain the sip code and proto_specific_hangup_cause
> will contain sip:<code>.
>
>
>   Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>  Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:
>
> My guess is that you will receive a message here:
>
> switch_status_t channel_receive_message(switch_core_session_t *session,
> switch_core_session_message_t *msg)
>
> The problem here is that you don't have the exact SIP code but there is a
> clear relationship between the codes and the messages you receive on the
> channel, so I am guessing that is all the same.
>
> Hope this helps.
>
> jmesquita
>
> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com>wrote:
>
>> Hi,
>>
>> I want to implement a module where freeSWITCH would try to bridge to an
>> extension and if the bridging operation fails, my module can use the hangup
>> code to determine the next cause of action.
>>
>> With switch_caller_extension_add_application(session, extension, "bridge",
>> "sofia/gateway/mygw/1232323);, if there is an error ( 503 received for
>> instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the
>> module's APP) and go on to the next action.  Is there anyway to control it
>> so that freeSWITCH would remain to be within the module's APP funtion and
>> continue executing the code after switch_call_extension_add_application,
>> when let's say a 4XX or 5XX or CANCEL ( from originator) is received?
>>
>> I have tried it and found that if the bridging is successful, freeSWITCH
>> would continue executing the code after
>> switch_caller_extension_add_application, but if an error is received, then
>> it would just move on to the next action.
>>
>> Does anyone know how to deal with this problem?
>>
>> Woody
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Wed Aug  5 17:36:56 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 5 Aug 2009 19:36:56 -0500
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
	<a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
Message-ID: <9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>

Hi,

You can set the "continue_on_fail" variable to true (or to the hangup  
causes you want it to ignore) and it'll keep executing whats queued.
For receive_message, unless you hook the session thats being created  
as a B-leg, you won't get anything relevant.
Also set hangup_after_bridge=true if you want to stop failing over  
when it worked.

Im curious, what are you coding? you can transfer the call in the  
dialplan without having to do all this manual queuing in C, thats why  
the routing state and dialplan modules exist. If you need to pull data  
from somewhere you can fill in channel variables that you can  
reference in the dialplan.

/*!
   \brief Transfer an existing session to another location
   \param session the session to transfer
   \param extension the new extension
   \param dialplan the new dialplan (OPTIONAL, may be NULL)
   \param context the new context (OPTIONAL, may be NULL)
*/
SWITCH_DECLARE(switch_status_t) switch_ivr_session_transfer(_In_  
switch_core_session_t *session, const char *extension, const char  
*dialplan,
															const char *context);



Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 5-Aug-09 um 7:20 PM schrieb Woody Dickson:

> Hi,
>
> The problem is that I need freeswitch to continue executing the code  
> after switch_status_t channel_receive_message even when it gets  
> error SIP code from the destination.  Is that possible?
>
> I know if I set up another action after my module in the  
> dialplan.xml, I can catch that.
>
> But I would like the code to execute within the route that I have.   
> Is that doable?
>
> Woody
>
> On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> The hangup cause will be in the originate_disposition channel  
> variable on the A-leg.
>
> sip_term_status will contain the sip code and  
> proto_specific_hangup_cause will contain sip:<code>.
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:
>
>> My guess is that you will receive a message here:
>>
>> switch_status_t channel_receive_message(switch_core_session_t  
>> *session, switch_core_session_message_t *msg)
>>
>> The problem here is that you don't have the exact SIP code but  
>> there is a clear relationship between the codes and the messages  
>> you receive on the channel, so I am guessing that is all the same.
>>
>> Hope this helps.
>>
>> jmesquita
>>
>> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com 
>> > wrote:
>> Hi,
>>
>> I want to implement a module where freeSWITCH would try to bridge  
>> to an extension and if the bridging operation fails, my module can  
>> use the hangup code to determine the next cause of action.
>>
>> With switch_caller_extension_add_application(session, extension,  
>> "bridge", "sofia/gateway/mygw/1232323);, if there is an error ( 503  
>> received for instance ) in the outgoing INVITE, freeSWITCH would  
>> leave my module ( or the module's APP) and go on to the next  
>> action.  Is there anyway to control it so that freeSWITCH would  
>> remain to be within the module's APP funtion and continue executing  
>> the code after switch_call_extension_add_application, when let's  
>> say a 4XX or 5XX or CANCEL ( from originator) is received?
>>
>> I have tried it and found that if the bridging is successful,  
>> freeSWITCH would continue executing the code after  
>> switch_caller_extension_add_application, but if an error is  
>> received, then it would just move on to the next action.
>>
>> Does anyone know how to deal with this problem?
>>
>> Woody
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vladrodionov at gmail.com  Wed Aug  5 16:30:09 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Wed, 5 Aug 2009 16:30:09 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to configure?)
Message-ID: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>

Hi, everybody

This is a newbie question: Suppose I have XX (variable dynamic number) DIDs
assigned to one sip trunk (from VOIP provider ABC ). All calls coming from
VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is it
possible in FS? If yes, how everything should be configuered? Dialplan, sip
gateway? One more question: suppose it is doeable as I hope then how can I
get in my script CalleeID (not a CallerID)? Basicaly,

I want to acomplish the following:

1. Avoid re-configuring FS every time I got new bunch of DIDs
assigned/released from/to my Voip provider.
2. Have a way of extracting CalleeID in my script.

TIA,

Vladimir Rodionov
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From dujinfang at gmail.com  Wed Aug  5 18:26:59 2009
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 6 Aug 2009 09:26:59 +0800
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
Message-ID: <23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>

mod_easyroute?

2009/8/6 Vladimir Rodionov <vladrodionov at gmail.com>

> Hi, everybody
>
> This is a newbie question: Suppose I have XX (variable dynamic number) DIDs
> assigned to one sip trunk (from VOIP provider ABC ). All calls coming from
> VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is it
> possible in FS? If yes, how everything should be configuered? Dialplan, sip
> gateway? One more question: suppose it is doeable as I hope then how can I
> get in my script CalleeID (not a CallerID)? Basicaly,
>
> I want to acomplish the following:
>
> 1. Avoid re-configuring FS every time I got new bunch of DIDs
> assigned/released from/to my Voip provider.
> 2. Have a way of extracting CalleeID in my script.
>
> TIA,
>
> Vladimir Rodionov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vladrodionov at gmail.com  Wed Aug  5 18:57:41 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Wed, 5 Aug 2009 18:57:41 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
	<23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
Message-ID: <3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>

No, it is more like static routing. I need my *script program* be invoked
when somebody dial in. That is it. One script for all inbound DIDs. Suppose
I have thousand of them.  I think I know how to accomplish this but I am not
sure yet.

in my dialplan I need to define:

<!-- Launch a JavaScript application if dialed in-->
   <extension name="ProviderABC">
    <condition field="source" expression="mod_sofia"/>
    *<condition field="destination_number" expression="^1NXXNXXXXXX$">*
     <action application="javascript"
data="/usr/local/freeswitch/scripts/myapp.js"/>
    </condition>
   </extension>


In provider configuration:

<gateway name="voicepulse">
           <!--/// account username *required* ///-->
           <param name="username" value="your-username"/>
           <!--/// auth realm: *optional* same as gateway name, if blank ///-->
           <param name="realm" value="nyc.voicepulse.com"/>
           <!--/// account password *required* ///-->
           <param name="password" value="your-password"/>
           <!--/// extension for inbound calls: *optional* same as
username, if blank ///-->
*           * *<param name="extension" value="1NXXNXXXXXX"/> *
           <!--/// proxy host: *optional* same as realm, if blank ///-->
           <param name="proxy" value="nyc.voicepulse.com"/>
           <!--/// expire in seconds: *optional* 3600, if blank ///-->
           <param name="expire-seconds" value="600"/>

	   <param name="register" value="true"/>
         </gateway>


Something like this, yes? I can use regular expressions in
destination_number?

Q: There is object Session in JavaScript, Lua. Is Session.destination ==
destination_number from incoming call? It is not clear for me from what I
have read so far.

TIA,

-Vladimir Rodionov

On Wed, Aug 5, 2009 at 6:26 PM, Seven Du <dujinfang at gmail.com> wrote:

> mod_easyroute?
>
> 2009/8/6 Vladimir Rodionov <vladrodionov at gmail.com>
>
>> Hi, everybody
>>
>> This is a newbie question: Suppose I have XX (variable dynamic number)
>> DIDs assigned to one sip trunk (from VOIP provider ABC ). All calls coming
>> from VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is
>> it possible in FS? If yes, how everything should be configuered? Dialplan,
>> sip gateway? One more question: suppose it is doeable as I hope then how can
>> I get in my script CalleeID (not a CallerID)? Basicaly,
>>
>> I want to acomplish the following:
>>
>> 1. Avoid re-configuring FS every time I got new bunch of DIDs
>> assigned/released from/to my Voip provider.
>> 2. Have a way of extracting CalleeID in my script.
>>
>> TIA,
>>
>> Vladimir Rodionov
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Aug  5 19:03:26 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 21:03:26 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
	<b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
Message-ID: <87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>

On Wed, Aug 5, 2009 at 6:44 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Well, I would randomly insert all of those cases to make it more
> realistic... only thing I cannot manage to issue USER_BUSY from lua (and
> neither from the dialplan, actually).
>
> <anti-action application="respond" data="407" /> (407 or 486 or
> whatever...)
>
>
> doesn't behave as I expected and neither
>
> <action application="hangup" data="407" /> (407 or 486 or USER_BUSY or
> whatever...)
>
>
> and I cannot find a a session:reject(<hangupcause>) method in lua.
>
> Can you give me a hint?
>

You can execute pretty much any dialplan app with the session:execute
command:
http://wiki.freeswitch.org/wiki/Lua#session:execute

Try something like:
session:execute("hangup","USER_BUSY");

-MC
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From msc at freeswitch.org  Wed Aug  5 20:40:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 22:40:31 -0500
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
	<23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
	<3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>
Message-ID: <87f2f3b90908052040s21545229qba24937bd2b14540@mail.gmail.com>

On Wed, Aug 5, 2009 at 8:57 PM, Vladimir Rodionov <vladrodionov at gmail.com>wrote:

> No, it is more like static routing. I need my *script program* be invoked
> when somebody dial in. That is it. One script for all inbound DIDs. Suppose
> I have thousand of them.  I think I know how to accomplish this but I am not
> sure yet.
>

Have the external profile be used only for provider ABC, or define a new
profile. Then in the profile have the calls go to a specific context. You
could have something like this in the sip profile definition:

<param name="context" value="abc_calls"/>

Then create a dialplan context called "abc_calls" that handles all inbound
calls. Create a file in conf/dialplan/ called abc_calls.xml:

<include>
  <context name="abc_calls">
    <extension name="abc_calls">
      <condition field="destination_number" expression="^(.*)$">
        <action application="lua" data="myscript.lua"/>
      </condition>
    </extension>
  </context>
</include>

Essentially you're just creating a SIP profile and a dialplan context that
are servicing your VoIP provider. You can add other profiles/contexts for
other providers if need be.

Let us know how it goes...
-MC
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From pete at privateconnect.com  Wed Aug  5 20:45:30 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Wed, 05 Aug 2009 20:45:30 -0700
Subject: [Freeswitch-users]
 =?utf-8?q?Multiple_DIDs_per_SIP_trunk_=28how_t?=
 =?utf-8?q?o_configure=3F=29?=
Message-ID: <20090805204530.2ad02225396a31c9de30536f2e338977.43419788fb.wbe@email04.secureserver.net>

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From woodydickson at gmail.com  Wed Aug  5 20:54:28 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 6 Aug 2009 11:54:28 +0800
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
	<a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
	<9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>
Message-ID: <a44688a0908052054j405c1707j382b3d6ccac8f8a9@mail.gmail.com>

Hi,

In my module, I will collect a list of available failover route that I can
use to failover to whenever a particular error is received.  However, these
available routes has different condition and the condition changes every
half a minute.  Therefore, I need to catch the hangup cause after bridge,
and then figure out the next workable available route based on the latest
condition setting.

It seems like this is only prossible to be done within a C module.  Any
suggestion will be greatly appreciated.

Woody

On Thu, Aug 6, 2009 at 8:36 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
> You can set the "continue_on_fail" variable to true (or to the hangup
> causes you want it to ignore) and it'll keep executing whats queued. For
> receive_message, unless you hook the session thats being created as a B-leg,
> you won't get anything relevant.
> Also set hangup_after_bridge=true if you want to stop failing over when it
> worked.
>
> Im curious, what are you coding? you can transfer the call in the dialplan
> without having to do all this manual queuing in C, thats why the routing
> state and dialplan modules exist. If you need to pull data from somewhere
> you can fill in channel variables that you can reference in the dialplan.
>
> /*!
>   \brief Transfer an existing session to another location
>   \param session the session to transfer
>   \param extension the new extension
>   \param dialplan the new dialplan (OPTIONAL, may be NULL)
>   \param context the new context (OPTIONAL, may be NULL)
> */
> SWITCH_DECLARE(switch_status_t) switch_ivr_session_transfer(_In_
> switch_core_session_t *session, const char *extension, const char *dialplan,
> const char *context);
>
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 5-Aug-09 um 7:20 PM schrieb Woody Dickson:
>
> Hi,
>
> The problem is that I need freeswitch to continue executing the code after
> switch_status_t channel_receive_message even when it gets error SIP code
> from the destination.  Is that possible?
>
> I know if I set up another action after my module in the dialplan.xml, I
> can catch that.
>
> But I would like the code to execute within the route that I have.  Is that
> doable?
>
> Woody
>
> On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
>> The hangup cause will be in the originate_disposition channel
>> variable on the A-leg.
>> sip_term_status will contain the sip code and proto_specific_hangup_cause
>> will contain sip:<code>.
>>
>>
>>   Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>>  Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:
>>
>> My guess is that you will receive a message here:
>>
>> switch_status_t channel_receive_message(switch_core_session_t *session,
>> switch_core_session_message_t *msg)
>>
>> The problem here is that you don't have the exact SIP code but there is a
>> clear relationship between the codes and the messages you receive on the
>> channel, so I am guessing that is all the same.
>>
>> Hope this helps.
>>
>> jmesquita
>>
>> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com>wrote:
>>
>>> Hi,
>>>
>>> I want to implement a module where freeSWITCH would try to bridge to an
>>> extension and if the bridging operation fails, my module can use the hangup
>>> code to determine the next cause of action.
>>>
>>> With switch_caller_extension_add_application(session, extension,
>>> "bridge", "sofia/gateway/mygw/1232323);, if there is an error ( 503 received
>>> for instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or
>>> the module's APP) and go on to the next action.  Is there anyway to control
>>> it so that freeSWITCH would remain to be within the module's APP funtion and
>>> continue executing the code after switch_call_extension_add_application,
>>> when let's say a 4XX or 5XX or CANCEL ( from originator) is received?
>>>
>>> I have tried it and found that if the bridging is successful, freeSWITCH
>>> would continue executing the code after
>>> switch_caller_extension_add_application, but if an error is received, then
>>> it would just move on to the next action.
>>>
>>> Does anyone know how to deal with this problem?
>>>
>>> Woody
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From velu.technical at gmail.com  Wed Aug  5 21:38:25 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Thu, 6 Aug 2009 10:08:25 +0530
Subject: [Freeswitch-users] Fwd: execute function in ESL.pm module is not
	working
In-Reply-To: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
References: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
Message-ID: <1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>

Please any one help for this problem..

---------- Forwarded message ----------
From: velusamy velu <velu.technical at gmail.com>
Date: Wed, Aug 5, 2009 at 11:44 AM
Subject: execute function in ESL.pm module is not working
To: freeswitch-users at lists.freeswitch.org


Dear All,
   I registered alarm signal in my Perl server program.
   If ALARM signal occurred I execute the following statement in signal
handler.
    "$conn->execute("playback",$sound_path."voicemail/vm-goodbye.wav")"
  The above statement didn't play that wave file. But before generating the
ALARM signal it worked.

What is the problem?

Please help me in this problem....

Also Is there any idea to do timeout for DTMF digits?

Thanks...

Regards,
K.Velusamy
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From msc at freeswitch.org  Wed Aug  5 22:54:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Aug 2009 00:54:17 -0500
Subject: [Freeswitch-users] Fwd: execute function in ESL.pm module is
	not working
In-Reply-To: <1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>
References: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
	<1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>
Message-ID: <87f2f3b90908052254l3ecc7fa0ybc92d87c587a9b0d@mail.gmail.com>

On Wed, Aug 5, 2009 at 11:38 PM, velusamy velu <velu.technical at gmail.com>wrote:

> Please any one help for this problem..
>
>
Sorry for the delay but many of the FreeSWITCH experts are at ClueCon right
now so we'll ask for your patience... in the meantime could you pastebin
your script and your dialplan entry so that we can take a look at them?

Thanks,
MC
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From mayamatakeshi at gmail.com  Thu Aug  6 00:03:06 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 6 Aug 2009 16:03:06 +0900
Subject: [Freeswitch-users] Monitoring On-Hold/Off-Hold
In-Reply-To: <5a8712120908051055k95f01d4x61ef7b7dff80298a@mail.gmail.com>
References: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>
	<5a8712120908051055k95f01d4x61ef7b7dff80298a@mail.gmail.com>
Message-ID: <15b9404e0908060003s2eb17434q1de769fdde444ad2@mail.gmail.com>

2009/8/6 Jo?o Mesquita <jmesquita at gmail.com>:
> I only see one way out of this. If you manage presence, an event like the
> following is sent:
>
> Event-Name: PRESENCE_IN
> Core-UUID: 189b12c0-7fb0-11de-b0bc-37eec03ad00f
> FreeSWITCH-Hostname: cl-t146-421cl
> FreeSWITCH-IPv4: XXXXXX
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-05%2013%3A42%3A24
> Event-Date-GMT: Wed,%2005%20Aug%202009%2017%3A42%3A24%20GMT
> Event-Date-Timestamp: 1249494144628132
> Event-Calling-File: switch_channel.c
> Event-Calling-Function: switch_channel_presence
> Event-Calling-Line-Number: 472
> Channel-State: CS_HIBERNATE
> Channel-State-Number: 8
> Channel-Name: XXXXX
> Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
> Call-Direction: inbound
> Presence-Call-Direction: inbound
> Answer-State: answered
> Caller-Username: 1000
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: Mesquita
> Caller-Caller-ID-Number: 1000
> Caller-Network-Addr: XXXXX
> Caller-Destination-Number: 1005
> Caller-Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
> Caller-Source: mod_sofia
> Caller-Context: XXXXX
> Caller-Channel-Name: XXXXX
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249494132128119
> Caller-Channel-Created-Time: 1249494132128119
> Caller-Channel-Answered-Time: 1249494139500129
> Caller-Channel-Progress-Time: 1249494132368119
> Caller-Channel-Progress-Media-Time: 0
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Other-Leg-Username: 1000
> Other-Leg-Dialplan: XML
> Other-Leg-Caller-ID-Name: Joao%20Mesquita
> Other-Leg-Caller-ID-Number: 1000
> Other-Leg-Network-Addr: 190.2.41.65
> Other-Leg-Destination-Number:
> sip%3A1005%40192.168.0.106%3A4559%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1005%2540190.2.41.65%253A4559
> Other-Leg-Unique-ID: 4e7622ac-81e7-11de-b0bc-37eec03ad00f
> Other-Leg-Source: mod_sofia
> Other-Leg-Context: XXXXX
> Other-Leg-Channel-Name: XXXXXX
> Other-Leg-Screen-Bit: true
> Other-Leg-Privacy-Hide-Name: false
> Other-Leg-Privacy-Hide-Number: false
> proto: src/switch_channel.c
> login: src/switch_channel.c
> from: XXXXXX
> rpid: unknown
> status: hold
> event_type: presence
> alt_event_type: dialog
> event_count: 3
>
> Content-Length: 543
> Content-Type: text/event-plain
>
> Other than that, I think it can be patched. I will take a look at it.

Thanks, that would be the best.

Just in case someone else needs this:
I have also tried to watch for
CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE with Application set to
playback and some indication of MOH in the Application-Data header.
That would work but:
- they will be fired continuously if you set hold_music=some_file
- they will not be fired if you set hold_music=silence (of course)


From enno.egbert at googlemail.com  Thu Aug  6 02:09:24 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Thu, 6 Aug 2009 02:09:24 -0700 (PDT)
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
References: <24823167.post@talk.nabble.com>
	<ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
Message-ID: <24841050.post@talk.nabble.com>


Hi Jim,

yes! It?s possible to call the IP200 full encrypted for example from a SNOM
or phonerlite. But when i try to call the SNOM from a innovaphone, the call
fails and i only hear the mailbox.

To modify the dialplan i am not so sure how i works. I don?t have any
experience of configure freeswitch or working with xml files. :confused:

In my dialplan i just modify a few lines. If you want, you can have a look
on the file in the attachment. 

http://www.nabble.com/file/p24841050/default.xml default.xml 

Thanks for your help. =)

NOx


Jim Burke-2 wrote:
> 
> Hi NOx,
> 
> Can you clarify the direction of the calls.  When you say outgoing do
> you mean a call is terminating to the ip200?
> 
> I have been down a similar path while testing Eyebeam.  If the
> terminating phone sets an option to only accept secure calls and FS
> does not send Secure Descriptions in the INVITE, Eyebeam would respond
> with 415 response code and the call would fail.  Depending on your
> diaplan this could send your call to voicemail.
> 
> To fix it I added the following code to dialplan.
> <action application="set" data="continue_on_fail=79"/>
> <bridge blah blah blah>
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="proxy_media=true"/>
> <action application="set" data="ringback=$${uk-ring}"/>
> <action application="pre_answer"/>
> <action application="export" data="sip_secure_media=true"/>
> <bridge blah blah blah>
> 
> The continue on fail captures the 415 response code forces the call to
> continue to the next bridge while sip_secure_media forces the second
> invite to include security descriptors.  The rest was required because
> I did not want to proxy media if the call was not secure, obviously if
> the call is secure on a point to point basis FS will have to proxy the
> media and this was the only way I could find for it to work.
> 
> Hope this helps.
> 
> Regards,
> 
> 
> On Wed, Aug 5, 2009 at 6:52 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>>
>> Hello,
>>
>> i have a problem using a innovaphone ip200 with freeswitch and tls/srtp.
>> The
>> freeswitch certificate is in the trust list of the phone and it works
>> with
>> tls for incomming calls. But outgoing calls were rejected to the mailbox.
>> The freeswitch configuration is ok, because it works with a snom 320.
>>
>> Who can help me to confugure the IP200?
>>
>> Thanks
>>
>> NOx
>> --
>> View this message in context:
>> http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24841050.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From juanbackson at gmail.com  Thu Aug  6 02:24:03 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 6 Aug 2009 17:24:03 +0800
Subject: [Freeswitch-users] Question about dynamic registration
In-Reply-To: <9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>
References: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
	<9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>
Message-ID: <27c25bc40908060224l38ad47fdje17065b13647905d@mail.gmail.com>

Hi,

Is there a sample module that I can take a look at on how to do that?
I don't understand how to get the registration request and how to pass back
auth result to freeswitch.

JB

On Mon, Aug 3, 2009 at 8:42 PM, Brian West <brian at freeswitch.org> wrote:

> You could build your own module to do it how ever you please.  But
> forking a script every time to auth is not very scalable.
>
> /b
>
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From markmorreny at gmail.com  Thu Aug  6 02:26:53 2009
From: markmorreny at gmail.com (mark morreny)
Date: Thu, 6 Aug 2009 17:26:53 +0800
Subject: [Freeswitch-users] question about latest version of mod_limit
Message-ID: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>

Hello,

I have the following setup in the dialplan.  Then, I fire up sipp to send
5calls/s and I expect to get limit-pass=false in most of the INFO output.
However, I am getting all "limit-pass=pass".

Does anyone know what is wrong with my dialplan?


 <context name="internal">

  <extension name="test-limit">

    <condition field="destination_number" expression="^9(.*)$">

        <action application="limit_hash_execute" data="192.168.1.102 6000
1/1 transfer to-next" />
        <action application="set" data="limit-pass=false"/>
        <action application="info" />
        <action application="hangup" />

    </condition>

  </extension>
    <extension name="to-next">
        <condition field="destination_number" expression="^to-next$">
        <action application="set" data="limit-pass=true"/>
        <action application="info" />
        <action application="hangup" />



    </condition>
    </extension>
</context>
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From enno.egbert at googlemail.com  Thu Aug  6 02:52:01 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Thu, 6 Aug 2009 02:52:01 -0700 (PDT)
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <DF06C882-055F-488D-B860-46643AF9B22B@freeswitch.org>
References: <24823167.post@talk.nabble.com>
	<DF06C882-055F-488D-B860-46643AF9B22B@freeswitch.org>
Message-ID: <24841078.post@talk.nabble.com>


Hi Brian, 

where i have to ship the phone? Maybe i can ship it to you. But i have first
to ask the owner, because i also borrow it.

Thanks for your help

NOx
 

Brian West-3 wrote:
> 
> Can you ship me a phone to test with?  That's usually the missing  
> element when testing this stuff is I just can't afford to buy every  
> phone to test with.
> 
> /b
> 
> On Aug 5, 2009, at 3:52 AM, NOx-WHV wrote:
> 
>>
>> Hello,
>>
>> i have a problem using a innovaphone ip200 with freeswitch and tls/ 
>> srtp. The
>> freeswitch certificate is in the trust list of the phone and it  
>> works with
>> tls for incomming calls. But outgoing calls were rejected to the  
>> mailbox.
>> The freeswitch configuration is ok, because it works with a snom 320.
>>
>> Who can help me to confugure the IP200?
>>
>> Thanks
>>
>> NOx
>> -- 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24841078.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From thangappan143 at gmail.com  Thu Aug  6 02:55:56 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Thu, 6 Aug 2009 15:25:56 +0530
Subject: [Freeswitch-users] IVR on Freeswitch
Message-ID: <7aa29e790908060255p36577d41hffe05ccc380c2540@mail.gmail.com>

Dear all,

   I am in the process of implementing IVR in Perl using event bound socket
on FreeSWITCH. I want to use all the functionality of IVR in my
implementation.

   I have seen the XML MACRO (default implementation) in the ivr.conf .xml
and demo/en/demo-ivr.xml .In that file I don't want to handle the
inter-digit time out and response timeout and all.I can just configure the
seconds.It will automatically works specified in the  <menu> tag.

I want to specify the menu definitions how they have specified in the XML
field in the ivr.conf.xml
Is there any way in Perl to do that?

I want to handle timeout,interdigittimeout  in Perl.Please help me?





-- 
Regards,
Thangappan.M
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From dujinfang at gmail.com  Thu Aug  6 02:58:59 2009
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 6 Aug 2009 17:58:59 +0800
Subject: [Freeswitch-users] A few questions about lua
Message-ID: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>

ALL-
I have a few questions when scripting lua. According to wiki, it is possible
to run looping forever lua scripts through start-up config or luarun.

1) Will the lua script stop when unload mod_lua? I experienced core dump
when unload mod_lua while there was a running lua script. Reported on jira.

2) How to stop a forever running lua script?  I stop it by listening a
CUSTOM event fired elsewhere. See code below. Is there any standard way like
luastop ?

3) Any way to show how many running lua scripts? luashow ?

4) It seems cannot get the lua script name in a lua script, I made a patch
to jira by assign it to the argv[0].

5) Seems that only EventConsumer("all") working.
EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
idea to this?

Thanks a lot.



code example:

con = freeswitch.EventConsumer("all");

argv[0] = "test.lua"

freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "] Starting
=====\n");

local all_events = 0


for e in (function() return con:pop(1) end) do
  -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
   all_events = all_events + 1;
freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
 event_name = e:getHeader("Event-Name") or ""
event_subclass = e:getHeader("Event-Subclass") or ""
 if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
"]---Exiting------\n")
  break
end

end
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From dome at tel.co.th  Thu Aug  6 03:48:58 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 6 Aug 2009 17:48:58 +0700
Subject: [Freeswitch-users] Numeric Value Ranges Expressions in dialplan
Message-ID: <8ccbff060908060348l75ee1062ua6145da2d6f7c4e9@mail.gmail.com>

Dear sir,

    Is posible to check numeric range in dialplan (expression).
   example i got balance vaiable from somewhere and want to check > 0
or not before call bridge application.
   ( I don't want to call scripts)

Best regards.

Dome C.


From jason at jasonjgw.net  Thu Aug  6 04:19:04 2009
From: jason at jasonjgw.net (Jason White)
Date: Thu, 6 Aug 2009 21:19:04 +1000
Subject: [Freeswitch-users] Numeric Value Ranges Expressions in dialplan
In-Reply-To: <8ccbff060908060348l75ee1062ua6145da2d6f7c4e9@mail.gmail.com>
References: <8ccbff060908060348l75ee1062ua6145da2d6f7c4e9@mail.gmail.com>
Message-ID: <20090806111903.GB17479@jdc.jasonjgw.net>

Dome Charoenyost <dome at tel.co.th> wrote:
>     Is posible to check numeric range in dialplan (expression).
>    example i got balance vaiable from somewhere and want to check > 0
> or not before call bridge application.
>    ( I don't want to call scripts)

Can you write a regular expression to match it?

^[1-9]\d*$
for example, might be a good start to identify non-zero integers.



From saeedahmad1981 at gmail.com  Thu Aug  6 05:05:48 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 6 Aug 2009 14:05:48 +0200
Subject: [Freeswitch-users] SVN error
Message-ID: <E47E8DBFC76D4E769B7BBF60561CAA27@saeedlaptop>

Hi,

While doing 'make current' or 'svn up' I am getting following errors:

svn: REPORT request failed on '/svn/!svn/vcc/default'

svn: Can't find a temporary directory: Internal error

 

- Saeed

 

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From raffaele.p.guidi at gmail.com  Thu Aug  6 05:49:19 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 6 Aug 2009 14:49:19 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
	<b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
	<87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>
Message-ID: <b0a8792f0908060549v40196e6fm6da7f95f47193bb4@mail.gmail.com>

Done, it (of course, thanks) worked smoothly. I've published the example on
the wiki.
http://wiki.freeswitch.org/wiki/Fakecall_responder (and linked in mod_lua
samples)

Regards,
   Raffaele

On Thu, Aug 6, 2009 at 04:03, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 6:44 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> Well, I would randomly insert all of those cases to make it more
>> realistic... only thing I cannot manage to issue USER_BUSY from lua (and
>> neither from the dialplan, actually).
>>
>> <anti-action application="respond" data="407" /> (407 or 486 or
>> whatever...)
>>
>>
>> doesn't behave as I expected and neither
>>
>> <action application="hangup" data="407" /> (407 or 486 or USER_BUSY or
>> whatever...)
>>
>>
>> and I cannot find a a session:reject(<hangupcause>) method in lua.
>>
>> Can you give me a hint?
>>
>
> You can execute pretty much any dialplan app with the session:execute
> command:
> http://wiki.freeswitch.org/wiki/Lua#session:execute
>
> Try something like:
> session:execute("hangup","USER_BUSY");
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From elihay at savion.huji.ac.il  Thu Aug  6 03:47:00 2009
From: elihay at savion.huji.ac.il (Eli Hayun)
Date: Thu, 06 Aug 2009 13:47:00 +0300
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
Message-ID: <1249555620.5449.1.camel@eli-desktop>

Hi
I dont know about events so much but I cannot see variable "e" is
setting

event_name = e:getHeader("Event-Name") or "" 
event_subclass = e:getHeader("Event-Subclass") or "" 

regurds
Eli

On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:

> ALL-
> 
> 
> 
> I have a few questions when scripting lua. According to wiki, it is
> possible to run looping forever lua scripts through start-up config or
> luarun. 
> 
> 
> 1) Will the lua script stop when unload mod_lua? I experienced core
> dump when unload mod_lua while there was a running lua script.
> Reported on jira.
> 
> 
> 2) How to stop a forever running lua script?  I stop it by listening a
> CUSTOM event fired elsewhere. See code below. Is there any standard
> way like luastop ?
> 
> 
> 3) Any way to show how many running lua scripts? luashow ?
> 
> 
> 4) It seems cannot get the lua script name in a lua script, I made a
> patch to jira by assign it to the argv[0].
> 
> 
> 5) Seems that only EventConsumer("all") working.
> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work.
> Any idea to this?
> 
> 
> Thanks a lot.
> 
> 
> 
> 
>                                                
> code example:
> 
> 
> con = freeswitch.EventConsumer("all");      
> 
> 
> argv[0] = "test.lua"
>                        
> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]
> Starting =====\n");
>                         
> local all_events = 0
> 
>                                      
> for e in (function() return con:pop(1) end) do
>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>    all_events = all_events + 1;
> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
> 
> event_name = e:getHeader("Event-Name") or ""
> event_subclass = e:getHeader("Event-Subclass") or ""
> 
> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
> "]---Exiting------\n")
>  break
> end
> 
> 
> end 
> 
> 
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From lakindia89 at gmail.com  Thu Aug  6 03:47:20 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 6 Aug 2009 16:17:20 +0530
Subject: [Freeswitch-users] Error while creating object
Message-ID: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>

Hi all,
Greets.

I am in the process of controlling the freeswitch with perl.
I have read about mod_perl and I wrote some scripts to test which works
fine.
Yesterday I tried to access the digit_set function.
So I create an object for the freeswitch::DTMF.
But it reported the following error.

2009-08-06 15:53:46 [ERR] mod_perl.c:69 Perl_safe_eval() [require
'/usr/local/freeswitch/conf/test.pl';]
No matching function for overloaded 'new_DTMF' at
/usr/local/freeswitch/perl/freeswitch.pm line 197.
Compilation failed in require at (eval 2) line 1.

Here is my code.

#!/usr/bin/perl
use strict;
use freeswitch;
our $session;
$session->execute("bridge","user/1010");
my $sess=&freeswitch::DTMF::new;
return 1;

The bridge is working fine. But while creating the object it said error.

Can any one explain why this happens and how can I correct it?
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From jim.page at redmatter.com  Thu Aug  6 04:24:15 2009
From: jim.page at redmatter.com (Jim Page)
Date: Thu, 6 Aug 2009 12:24:15 +0100
Subject: [Freeswitch-users] CURL directory issue
Message-ID: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>

Afternoon All

I wonder if someone (perhaps even the illustrious intralanman) could help me out with a problem I am experiencing with a CURL directory.

In the interests of understanding how the mechanism works, I am using a super-braindead php script to return info about a specific set of users. I plan to move to something more sophisticated once the proof of concept is complete, possibly based on intralanman's scripts.

The basic problem is that all works fine (boot, register, voicemail etc), except that user's <variables> seem not to be being read correctly, eg 'toll_allow' and 'user_context'. Here's a typical user XML message I am returning:

<document type="freeswitch/xml">
  <section name="directory">
   <domain name="pbx.redmatter.com">
    <params>
      <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
    </params>

    <variables>
      <variable name="record_stereo" value="true"/>
      <variable name="default_gateway" value="$${default_provider}"/>
      <variable name="default_areacode" value="$${default_areacode}"/>
      <variable name="transfer_fallback_extension" value="operator"/>
    </variables>

    <groups>
      <group name="default">
        <users>            <user id="1009" cidr="172.30.99.0/24">
               <params>
                 <param name="password" value="$${default_password}"/>
                 <param name="vm-password" value="1009"/>
                 <param name="vm-email-all-messages" value="true"/>
                 <param name="vm-attach-file" value="true"/>
                 <param name="vm-mailto" value="1009 at redmatter.com"/>
                 <param name="vm-keep-local-after-email" value="true"/>
                 <param name="vm_message_ext" value="mp3"/>
               </params>
               <variables>
                 <variable name="toll_allow" value="domestic,international,local"/>
                 <variable name="accountcode" value="1009"/>
                 <variable name="user_context" value="default"/>
                 <variable name="effective_caller_id_name" value="Extension 1009"/>
                 <variable name="effective_caller_id_number" value="1009"/>
                 <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
                 <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
                 <variable name="callgroup" value="techsupport"/>
               </variables>
             </user>        </users>
      </group>
    </groups>
   </domain>
 </section>
</document>

I return this kind of message in all cases except the (sip_auth_method=="REGISTER") request message where I return

<document type="freeswitch/xml">
  <section name="directory">
    <domain name="pbx.redmatter.com">
      <user id="1007">
        <params>
          <param name="password" value="1234"/>
        </params>
      </user>
    </domain>
  </section>
</document>

Also it's probably worth mentioning that I have removed all trace of xml from conf/directory and I don't believe there is a conflict happening there. 

The phones register correctly. The trouble is they don't operate on the correct dialplan context (I fixed that by hardcoding the internal gateway to dialplan default), but the 'toll_allow' variable is now not working so that outbound calls fail, which is what made me think that the user variables are being ignored.

Freeswitch version is 1.0.4, built by me and running on a dell 1950 running Centos 5.3 x86_64. HTTP application is running on standard Centos 5.3 apache/php.

Any ideas gratefully and humbly received.

All the best
Jim


From kevin at johnnyvoip.com  Thu Aug  6 08:48:08 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Thu, 6 Aug 2009 11:48:08 -0400
Subject: [Freeswitch-users] CURL directory issue
In-Reply-To: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>
References: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>
Message-ID: <ab57a3e80908060848x64bbc808r162273eb4cbcd363@mail.gmail.com>

Try returning the full information on the register. It may be that the
variables are read onto the user profile upon registration and since you are
only supplying a dumbed down version for registration the variables aren't
being read and cached.

Regards,
   Kevin Green

On Thu, Aug 6, 2009 at 7:24 AM, Jim Page <jim.page at redmatter.com> wrote:

> Afternoon All
>
> I wonder if someone (perhaps even the illustrious intralanman) could help
> me out with a problem I am experiencing with a CURL directory.
>
> In the interests of understanding how the mechanism works, I am using a
> super-braindead php script to return info about a specific set of users. I
> plan to move to something more sophisticated once the proof of concept is
> complete, possibly based on intralanman's scripts.
>
> The basic problem is that all works fine (boot, register, voicemail etc),
> except that user's <variables> seem not to be being read correctly, eg
> 'toll_allow' and 'user_context'. Here's a typical user XML message I am
> returning:
>
> <document type="freeswitch/xml">
>  <section name="directory">
>   <domain name="pbx.redmatter.com">
>    <params>
>      <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>    </params>
>
>    <variables>
>      <variable name="record_stereo" value="true"/>
>      <variable name="default_gateway" value="$${default_provider}"/>
>      <variable name="default_areacode" value="$${default_areacode}"/>
>      <variable name="transfer_fallback_extension" value="operator"/>
>    </variables>
>
>    <groups>
>      <group name="default">
>        <users>            <user id="1009" cidr="172.30.99.0/24">
>               <params>
>                 <param name="password" value="$${default_password}"/>
>                 <param name="vm-password" value="1009"/>
>                 <param name="vm-email-all-messages" value="true"/>
>                 <param name="vm-attach-file" value="true"/>
>                 <param name="vm-mailto" value="1009 at redmatter.com"/>
>                 <param name="vm-keep-local-after-email" value="true"/>
>                 <param name="vm_message_ext" value="mp3"/>
>               </params>
>               <variables>
>                 <variable name="toll_allow"
> value="domestic,international,local"/>
>                 <variable name="accountcode" value="1009"/>
>                 <variable name="user_context" value="default"/>
>                 <variable name="effective_caller_id_name" value="Extension
> 1009"/>
>                 <variable name="effective_caller_id_number" value="1009"/>
>                 <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>                 <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>                 <variable name="callgroup" value="techsupport"/>
>               </variables>
>             </user>        </users>
>      </group>
>    </groups>
>   </domain>
>  </section>
> </document>
>
> I return this kind of message in all cases except the
> (sip_auth_method=="REGISTER") request message where I return
>
> <document type="freeswitch/xml">
>  <section name="directory">
>    <domain name="pbx.redmatter.com">
>      <user id="1007">
>        <params>
>          <param name="password" value="1234"/>
>        </params>
>      </user>
>    </domain>
>  </section>
> </document>
>
> Also it's probably worth mentioning that I have removed all trace of xml
> from conf/directory and I don't believe there is a conflict happening there.
>
> The phones register correctly. The trouble is they don't operate on the
> correct dialplan context (I fixed that by hardcoding the internal gateway to
> dialplan default), but the 'toll_allow' variable is now not working so that
> outbound calls fail, which is what made me think that the user variables are
> being ignored.
>
> Freeswitch version is 1.0.4, built by me and running on a dell 1950 running
> Centos 5.3 x86_64. HTTP application is running on standard Centos 5.3
> apache/php.
>
> Any ideas gratefully and humbly received.
>
> All the best
> Jim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Thu Aug  6 08:52:07 2009
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 6 Aug 2009 23:52:07 +0800
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <1249555620.5449.1.camel@eli-desktop>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
	<1249555620.5449.1.camel@eli-desktop>
Message-ID: <EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>

for e in (function() return con:pop(1) end) do

btw, the script works.

Thanks.
On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
> Hi
> I dont know about events so much but I cannot see variable "e" is  
> setting
>
> event_name = e:getHeader("Event-Name") or ""
> event_subclass = e:getHeader("Event-Subclass") or ""
>
> regurds
> Eli
>
> On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
>> ALL-
>>
>>
>> I have a few questions when scripting lua. According to wiki, it is  
>> possible to run looping forever lua scripts through start-up config  
>> or luarun.
>>
>>
>> 1) Will the lua script stop when unload mod_lua? I experienced core  
>> dump when unload mod_lua while there was a running lua script.  
>> Reported on jira.
>>
>>
>> 2) How to stop a forever running lua script?  I stop it by  
>> listening a CUSTOM event fired elsewhere. See code below. Is there  
>> any standard way like luastop ?
>>
>>
>> 3) Any way to show how many running lua scripts? luashow ?
>>
>>
>> 4) It seems cannot get the lua script name in a lua script, I made  
>> a patch to jira by assign it to the argv[0].
>>
>>
>> 5) Seems that only EventConsumer("all") working.  
>> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to  
>> work. Any idea to this?
>>
>>
>> Thanks a lot.
>>
>>
>>
>>
>>
>> code example:
>>
>>
>> con = freeswitch.EventConsumer("all");
>>
>>
>> argv[0] = "test.lua"
>>
>> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]  
>> Starting =====\n");
>>
>> local all_events = 0
>>
>> for e in (function() return con:pop(1) end) do
>>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>>    all_events = all_events + 1;
>> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
>>
>> event_name = e:getHeader("Event-Name") or ""
>> event_subclass = e:getHeader("Event-Subclass") or ""
>>
>> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..  
>> "]---Exiting------\n")
>>  break
>> end
>>
>>
>> end
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ivan at myrvold.org  Thu Aug  6 08:53:35 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Thu, 6 Aug 2009 17:53:35 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
Message-ID: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>

Is skypiax now working on Mac OS X in Freeswitch?

Ivan


From brian at freeswitch.org  Thu Aug  6 08:55:37 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 6 Aug 2009 10:55:37 -0500
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
Message-ID: <1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>

I'm not sure about that one.... I haven't tried lately because the API  
differs on the Mac last I looked at it.

/b

On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:

> Is skypiax now working on Mac OS X in Freeswitch?
>
> Ivan



From vladrodionov at gmail.com  Thu Aug  6 08:59:36 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Thu, 6 Aug 2009 08:59:36 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <20090805204530.2ad02225396a31c9de30536f2e338977.43419788fb.wbe@email04.secureserver.net>
References: <20090805204530.2ad02225396a31c9de30536f2e338977.43419788fb.wbe@email04.secureserver.net>
Message-ID: <3c233920908060859q1b8bf67dkbe5750591446fd6@mail.gmail.com>

Pete,

Thank you for script. I can not find find channel variables rdnis,
sip_to_user and all others which start with "sb" on wiki page

http://wiki.freeswitch.org/wiki/Channel_Variables

Are they undocumented?

-Vladimir Rodionov


On Wed, Aug 5, 2009 at 8:45 PM, Pete Mueller <pete at privateconnect.com>wrote:

> Disclaimer: I'm not familiar with all the mods of FS, There may be one that
> does this already.  There are probably many ways to do this, I am just
> offering one that works well for me.
>
> Item #1 - Findout the callee #.   "destination_number" can be set to
> several different things based on the gateway configuration (forced override
> with an extension) and may or may not start with a "+" so the example below
> may not work.  To make matters worse, different gateways set fields
> differently when they hand off the call.  The most reliable I've found is
> "rdnis" or "sip_to_user" , however if you know you are going to stay with
> one gateway, you can relay on the oddities of the way they are configured.
> I had to write something relatively generic, so I moved all processing to a
> script (see #3 below)
>
> Item #2 - Find the caller ID. This is located in "caller_id_number", but
> remember in your processing that caller ID may be "anonymous", "restricted",
> "unknown" or some other word when dealing with blocked/private numbers.  You
> cannot looks for just numbers.
>
> Item #3 - Routing.  As I mentioned I have 100s of numbers across many
> gateways, so I needed a way to route the calls to the right places AND know
> which gateway the call came in on, so I can bridge the call out the same
> gateway.  I handled this by creating a small DB table (using postgreSQL) and
> connecting using LUA and luasql.  The table has three fields: number,
> gateway, and extension to route to.  In my public.xml I list all the places
> a call can be routed to and the last entry is a unconditional transfer to
> the "switchboard" script.  The switchboard script matches "rdnis" and
> "sip_to_user" to find the callee and then performs a lookup for the
> extension to route to.
>
> If you would like a copy of my switchboard script I can provide it to you
> in a PM.
> -pete
>
>  -------- Original Message --------
> Subject: Re: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
> configure?)
> From: Vladimir Rodionov <vladrodionov at gmail.com>
> Date: Wed, August 05, 2009 6:57 pm
> To: freeswitch-users at lists.freeswitch.org
>
> No, it is more like static routing. I need my *script program* be invoked
> when somebody dial in. That is it. One script for all inbound DIDs. Suppose
> I have thousand of them.  I think I know how to accomplish this but I am not
> sure yet.
>
> in my dialplan I need to define:
>
> <!-- Launch a JavaScript application if dialed in-->
>    <extension name="ProviderABC">
>     <condition field="source" expression="mod_sofia"/>
>     *<condition field="destination_number" expression="^1NXXNXXXXXX$">*
>
>      <action application="javascript" data="/usr/local/freeswitch/scripts/myapp.js"/>
>     </condition>
>    </extension>
>
>
> In provider configuration:
>
> <gateway name="voicepulse">
>            <!--/// account username *required* ///-->
>            <param name="username" value="your-username"/>
>            <!--/// auth realm: *optional* same as gateway name, if blank ///-->
>
>            <param name="realm" value="nyc.voicepulse.com"/>
>            <!--/// account password *required* ///-->
>            <param name="password" value="your-password"/>
>
>            <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
> *           * *<param name="extension" value="1NXXNXXXXXX"/> *
>            <!--/// proxy host: *optional* same as realm, if blank ///-->
>
>            <param name="proxy" value="nyc.voicepulse.com"/>
>            <!--/// expire in seconds: *optional* 3600, if blank ///-->
>            <param name="expire-seconds" value="600"/>
>
> 	   <param name="register" value="true"/>
>          </gateway>
>
>
> Something like this, yes? I can use regular expressions in
> destination_number?
>
> Q: There is object Session in JavaScript, Lua. Is Session.destination ==
> destination_number from incoming call? It is not clear for me from what I
> have read so far.
>
> TIA,
>
> -Vladimir Rodionov
>
> On Wed, Aug 5, 2009 at 6:26 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> mod_easyroute?
>>
>> 2009/8/6 Vladimir Rodionov <vladrodionov at gmail.com>
>>
>>>  Hi, everybody
>>>
>>> This is a newbie question: Suppose I have XX (variable dynamic number)
>>> DIDs assigned to one sip trunk (from VOIP provider ABC ). All calls coming
>>> from VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is
>>> it possible in FS? If yes, how everything should be configuered? Dialplan,
>>> sip gateway? One more question: suppose it is doeable as I hope then how can
>>> I get in my script CalleeID (not a CallerID)? Basicaly,
>>>
>>> I want to acomplish the following:
>>>
>>> 1. Avoid re-configuring FS every time I got new bunch of DIDs
>>> assigned/released from/to my Voip provider.
>>> 2. Have a way of extracting CalleeID in my script.
>>>
>>> TIA,
>>>
>>> Vladimir Rodionov
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vladrodionov at gmail.com  Thu Aug  6 09:00:36 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Thu, 6 Aug 2009 09:00:36 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <87f2f3b90908052040s21545229qba24937bd2b14540@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
	<23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
	<3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>
	<87f2f3b90908052040s21545229qba24937bd2b14540@mail.gmail.com>
Message-ID: <3c233920908060900j33cf4cb3hf50ca4452d1b245e@mail.gmail.com>

Thanks, I will give it it a try and let you know.

On Wed, Aug 5, 2009 at 8:40 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 8:57 PM, Vladimir Rodionov <vladrodionov at gmail.com>wrote:
>
>> No, it is more like static routing. I need my *script program* be invoked
>> when somebody dial in. That is it. One script for all inbound DIDs. Suppose
>> I have thousand of them.  I think I know how to accomplish this but I am not
>> sure yet.
>>
>
> Have the external profile be used only for provider ABC, or define a new
> profile. Then in the profile have the calls go to a specific context. You
> could have something like this in the sip profile definition:
>
> <param name="context" value="abc_calls"/>
>
> Then create a dialplan context called "abc_calls" that handles all inbound
> calls. Create a file in conf/dialplan/ called abc_calls.xml:
>
> <include>
>   <context name="abc_calls">
>     <extension name="abc_calls">
>       <condition field="destination_number" expression="^(.*)$">
>         <action application="lua" data="myscript.lua"/>
>       </condition>
>     </extension>
>   </context>
> </include>
>
> Essentially you're just creating a SIP profile and a dialplan context that
> are servicing your VoIP provider. You can add other profiles/contexts for
> other providers if need be.
>
> Let us know how it goes...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Thu Aug  6 09:37:25 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 6 Aug 2009 18:37:25 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org> 
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
Message-ID: <7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>

No, it needs implementation of the message pump between the module and
the Skype API.

It's probably kind of trivial, if no other problems I'm not aware of.

I do not have a Mac to implement it, tough :-(.

-giovanni





Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org> wrote:
> I'm not sure about that one.... I haven't tried lately because the API
> differs on the Mac last I looked at it.
>
> /b
>
> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>
>> Is skypiax now working on Mac OS X in Freeswitch?
>>
>> Ivan
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From nicolas at medularis.com  Thu Aug  6 09:38:34 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 6 Aug 2009 12:38:34 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
Message-ID: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>

I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:

2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL

But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
NORMAL_CLEARING. And the variable_originate_disposition has a value of
"failure". Where can I get the detail of the call/bridge failure due to
'ORIGINATOR_CANCEL' as reported through the console?

Thanks!

Nicolas
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From msc at freeswitch.org  Thu Aug  6 09:45:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Aug 2009 11:45:33 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <b0a8792f0908060549v40196e6fm6da7f95f47193bb4@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
	<b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
	<87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>
	<b0a8792f0908060549v40196e6fm6da7f95f47193bb4@mail.gmail.com>
Message-ID: <87f2f3b90908060945y5cb7e842le9c9ba02bc3c303b@mail.gmail.com>

On Thu, Aug 6, 2009 at 7:49 AM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Done, it (of course, thanks) worked smoothly. I've published the example on
> the wiki.
> http://wiki.freeswitch.org/wiki/Fakecall_responder (and linked in mod_lua
> samples)
>
> Regards,
>    Raffaele
>

Thanks for paying the wiki tax! We appreciate it when folks document their
knowledge. Please let me know if you have any wiki questions in the future.
-MC
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From vkozak at abisoft.spb.ru  Thu Aug  6 10:01:09 2009
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Thu, 6 Aug 2009 21:01:09 +0400
Subject: [Freeswitch-users] FreeSwitch doesn't play music on hold forbriged
	channel
Message-ID: <EC1055E8A337451F9F5ADC7414BDFDD9@abisoft.biz>


The scenario is the following:
FS User A dial an extension
Extention opens outbound socket channel to my application
My application bridges the call to FS User B
The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
User B is placed on hold but no music on hold is played to the caller (User A)


I have outbound socket channel and the following sequence of commands/event:
listening on [any] 8084 ...
connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
connect

myevents

SendMsg
call-command: execute
execute-app-name: bridge
execute-app-arg:user/1000 at uat.agent.starpoundtech.net

Channel-Username: 1001
Channel-Dialplan: XML
Channel-Caller-ID-Name: 1001
Channel-Caller-ID-Number: 1001
Channel-Network-Addr: 172.26.10.39
Channel-Destination-Number: 6666
Channel-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Channel-Source: mod_sofia
Channel-Context: default
Channel-Channel-Name: sofia/internal/1001%40172.26.200.250
Channel-Profile-Index: 1
Channel-Profile-Created-Time: 1249142681680114
Channel-Channel-Created-Time: 1249142681680114
Channel-Channel-Answered-Time: 0
Channel-Channel-Progress-Time: 0
Channel-Channel-Progress-Media-Time: 1249142681809352
Channel-Channel-Hangup-Time: 0
Channel-Channel-Transfer-Time: 0
Channel-Screen-Bit: true
Channel-Privacy-Hide-Name: false
Channel-Privacy-Hide-Number: false
Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: early
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 0
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
variable_sip_received_ip: 172.26.10.39
variable_sip_received_port: 13488
variable_sip_via_protocol: udp
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_sip_auth_realm: 172.26.200.250
variable_mailbox: 1001
variable_toll_allow: domestic,international,local
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_outbound_caller_id_name: StarPound%20FreeSWITCH
variable_outbound_caller_id_number: 0000000000
variable_callgroup: techsupport
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%40172.26.200.250
variable_sip_from_host: 172.26.200.250
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: bd11f93c
variable_sofia_profile_name: internal
variable_sip_req_user: 6666
variable_sip_req_uri: 6666%40172.26.200.250
variable_sip_req_host: 172.26.200.250
variable_sip_to_user: 6666
variable_sip_to_uri: 6666%40172.26.200.250
variable_sip_to_host: 172.26.200.250
variable_sip_contact_user: 1001
variable_sip_contact_port: 13488
variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
variable_sip_contact_host: 172.26.10.39
variable_channel_name: sofia/internal/1001%40172.26.200.250
variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
variable_sip_via_host: 172.26.10.39
variable_sip_via_port: 13488
variable_sip_via_rport: 13488
variable_max_forwards: 70
variable_presence_id: 1001%40172.26.200.250
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_remote_media_ip: 172.26.10.39
variable_remote_media_port: 29826
variable_read_codec: PCMU
variable_read_rate: 8000
variable_write_codec: PCMU
variable_write_rate: 8000
variable_use_profile: nat
variable_record_stereo: true
variable_transfer_fallback_extension: operator
variable_numbering_plan: US
variable_default_areacode: 918
variable_default_gateway: example.com
variable_user_name: default
variable_domain_name: 172.26.200.250
variable_current_application_data: 172.26.200.251%3A8084%20async%20full
variable_current_application: socket
variable_socket_host: 172.26.200.251
variable_local_media_ip: 172.26.200.250
variable_local_media_port: 29370
variable_endpoint_disposition: EARLY%20MEDIA
variable_sip_nat_detected: true
Content-Type: command/reply
Reply-Text: %2BOK%0A
Socket-Mode: async
Control: full

Content-Type: command/reply
Reply-Text: +OK Events Enabled

Content-Type: command/reply
Reply-Text: +OK

Content-Length: 1541
Content-Type: text/event-plain

Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: early
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 0
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Application: bridge
Application-Data: user/1000%40uat.agent.starpoundtech.net
Event-Name: CHANNEL_EXECUTE
Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
FreeSWITCH-Hostname: centos4-4-vm
FreeSWITCH-IPv4: 172.26.200.250
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-01%2020%3A04%3A51
Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A51%20GMT
Event-Date-Timestamp: 1249142691754598
Event-Calling-File: switch_core_session.c
Event-Calling-Function: switch_core_session_exec
Event-Calling-Line-Number: 1333

Content-Length: 5242
Content-Type: text/event-plain

Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: answered
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 1249142692414509
Caller-Channel-Progress-Time: 1249142691898434
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Leg-Username: 1001
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: Extension%201001
Other-Leg-Caller-ID-Number: 1001
Other-Leg-Network-Addr: 172.26.10.39
Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
Other-Leg-Source: mod_sofia
Other-Leg-Context: default
Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false
variable_sip_received_ip: 172.26.10.39
variable_sip_received_port: 13488
variable_sip_via_protocol: udp
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_sip_auth_realm: 172.26.200.250
variable_mailbox: 1001
variable_toll_allow: domestic,international,local
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_outbound_caller_id_name: StarPound%20FreeSWITCH
variable_outbound_caller_id_number: 0000000000
variable_callgroup: techsupport
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%40172.26.200.250
variable_sip_from_host: 172.26.200.250
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: bd11f93c
variable_sofia_profile_name: internal
variable_sip_req_user: 6666
variable_sip_req_uri: 6666%40172.26.200.250
variable_sip_req_host: 172.26.200.250
variable_sip_to_user: 6666
variable_sip_to_uri: 6666%40172.26.200.250
variable_sip_to_host: 172.26.200.250
variable_sip_contact_user: 1001
variable_sip_contact_port: 13488
variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
variable_sip_contact_host: 172.26.10.39
variable_channel_name: sofia/internal/1001%40172.26.200.250
variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
variable_sip_via_host: 172.26.10.39
variable_sip_via_port: 13488
variable_sip_via_rport: 13488
variable_max_forwards: 70
variable_presence_id: 1001%40172.26.200.250
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_remote_media_ip: 172.26.10.39
variable_remote_media_port: 29826
variable_read_codec: PCMU
variable_read_rate: 8000
variable_write_codec: PCMU
variable_write_rate: 8000
variable_use_profile: nat
variable_record_stereo: true
variable_transfer_fallback_extension: operator
variable_numbering_plan: US
variable_default_areacode: 918
variable_default_gateway: example.com
variable_user_name: default
variable_domain_name: 172.26.200.250
variable_socket_host: 172.26.200.251
variable_local_media_ip: 172.26.200.250
variable_local_media_port: 29370
variable_endpoint_disposition: EARLY%20MEDIA
variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
variable_current_application: bridge
variable_dialed_user: 1000
variable_dialed_domain: uat.agent.starpoundtech.net
variable_originate_disposition: failure
variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
variable_sip_redirect_contact_user_0: 1000
variable_sip_redirect_contact_host_0: 172.26.10.39
variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_sip_nat_detected: true
Event-Name: CHANNEL_ANSWER
Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
FreeSWITCH-Hostname: centos4-4-vm
FreeSWITCH-IPv4: 172.26.200.250
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-01%2020%3A04%3A52
Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
Event-Date-Timestamp: 1249142692414509
Event-Calling-File: switch_channel.c
Event-Calling-Function: switch_channel_perform_mark_answered
Event-Calling-Line-Number: 1776

Content-Length: 5233
Content-Type: text/event-plain

Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: answered
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 1249142692414509
Caller-Channel-Progress-Time: 1249142691898434
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Leg-Username: 1001
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: Extension%201001
Other-Leg-Caller-ID-Number: 1001
Other-Leg-Network-Addr: 172.26.10.39
Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
Other-Leg-Source: mod_sofia
Other-Leg-Context: default
Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false
variable_sip_received_ip: 172.26.10.39
variable_sip_received_port: 13488
variable_sip_via_protocol: udp
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_sip_auth_realm: 172.26.200.250
variable_mailbox: 1001
variable_toll_allow: domestic,international,local
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_outbound_caller_id_name: StarPound%20FreeSWITCH
variable_outbound_caller_id_number: 0000000000
variable_callgroup: techsupport
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%40172.26.200.250
variable_sip_from_host: 172.26.200.250
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: bd11f93c
variable_sofia_profile_name: internal
variable_sip_req_user: 6666
variable_sip_req_uri: 6666%40172.26.200.250
variable_sip_req_host: 172.26.200.250
variable_sip_to_user: 6666
variable_sip_to_uri: 6666%40172.26.200.250
variable_sip_to_host: 172.26.200.250
variable_sip_contact_user: 1001
variable_sip_contact_port: 13488
variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
variable_sip_contact_host: 172.26.10.39
variable_channel_name: sofia/internal/1001%40172.26.200.250
variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
variable_sip_via_host: 172.26.10.39
variable_sip_via_port: 13488
variable_sip_via_rport: 13488
variable_max_forwards: 70
variable_presence_id: 1001%40172.26.200.250
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_remote_media_ip: 172.26.10.39
variable_remote_media_port: 29826
variable_read_codec: PCMU
variable_read_rate: 8000
variable_write_codec: PCMU
variable_write_rate: 8000
variable_use_profile: nat
variable_record_stereo: true
variable_transfer_fallback_extension: operator
variable_numbering_plan: US
variable_default_areacode: 918
variable_default_gateway: example.com
variable_user_name: default
variable_domain_name: 172.26.200.250
variable_socket_host: 172.26.200.251
variable_local_media_ip: 172.26.200.250
variable_local_media_port: 29370
variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
variable_current_application: bridge
variable_dialed_user: 1000
variable_dialed_domain: uat.agent.starpoundtech.net
variable_sip_redirect_contact_user_0: 1000
variable_sip_redirect_contact_host_0: 172.26.10.39
variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_sip_nat_detected: true
variable_endpoint_disposition: ANSWER
variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
variable_originate_disposition: SUCCESS
Event-Name: CHANNEL_BRIDGE
Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
FreeSWITCH-Hostname: centos4-4-vm
FreeSWITCH-IPv4: 172.26.200.250
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-01%2020%3A04%3A52
Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
Event-Date-Timestamp: 1249142692414509
Event-Calling-File: switch_ivr_bridge.c
Event-Calling-Function: switch_ivr_multi_threaded_bridge
Event-Calling-Line-Number: 828

SendMsg 94b59a38-57c4-4703-9c6e-9985d832d119
call-command: execute
execute-app-name: hold

Content-Type: command/reply
Reply-Text: +OK




- I don't see the variable hold_music ... did you remove it? 

- I didn't. Moreover, I tried to set it explicitly using api uuid_setvar.
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From msc at freeswitch.org  Thu Aug  6 10:30:09 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Aug 2009 12:30:09 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 Release Announcement
Message-ID: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com>

We are happy to announce the official release of FreeSWITCH 1.0.4! Please
visit this link <http://digg.com/d3zt5X> to Digg and read the story, and
then spread the word!

Thanks for being such a great community!
-Michael
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From mattdfong at gmail.com  Thu Aug  6 11:25:32 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Thu, 6 Aug 2009 11:25:32 -0700
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
Message-ID: <4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>

Hi Nicolas,
do you have a copy of the .js code you can paste. I would guess tho, that
ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
false. Just a guess tho.

Hangup causes can be found here:
http://wiki.freeswitch.org/wiki/Hangup_causes

<http://wiki.freeswitch.org/wiki/Hangup_causes>--matt
hello hunter - hosted predictive dialer & voice broadcasting
http://www.hellohunter.com


On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> I'm bridging 2 calls in a javascript file, I originate the first call and
> then execute a bridge with an origination string for the second call. If I
> hangup the first call while trying to make the second call, I get this on
> the console:
>
> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/005622170039 [KILL]
> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/005622170039 [BREAK]
> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
> Cause: ORIGINATOR_CANCEL
>
> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
> NORMAL_CLEARING. And the variable_originate_disposition has a value of
> "failure". Where can I get the detail of the call/bridge failure due to
> 'ORIGINATOR_CANCEL' as reported through the console?
>
> Thanks!
>
> Nicolas
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Thu Aug  6 12:45:01 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 6 Aug 2009 15:45:01 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
Message-ID: <1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>

Hi Matt,

Actually I'm explicitly setting hangup_after_bridge to true, think setting
it to false would help? I'm going to try that.

Here's the JS code:
(Note: session.getVariable() doesn't work, FS complains saying it is not a
function, also tried self.session.getVariable() - that's what the wiki says
- and FS complains that self does not exist)

----------------
var uuid = argv[0]; // Call identifier
var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
var greeting_snd = "/var/audio/alert.wav";

console_log("notice", "*********** STARTING C2C Call ***********\n");
timeout = 30;

console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");

//var stUsRing = session.getVariable("us-ring");  // This doesn't work,
self.session.getVariable doesn't work either
var stUsRing = "%(2000,4000,440,480)";

// Create new_session
new_session = new Session(originate_str1);
console_log("notice", "*********** Leg1: " + new_session.cause + "
***********\n");

if (new_session.ready()) {
        // log to the console
        console_log("notice", "*********** Leg1 ("+dialstr1+") CONNECTED!
***********\n");
        console_log("notice", "*********** Playing greeting sound:
"+greeting_snd+" ***********\n");

        new_session.execute("sleep", 100);
        new_session.execute("playback", greeting_snd);

        // Originate second call and bridge
    originate_str2 =
"{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;

        // Create new_session
        new_session.execute("bridge", originate_str2);
        console_log("notice", "*********** Leg2: " + new_session.cause + "
***********\n");

        if (new_session.ready()) {
                console_log("notice", "*********** Leg2 ("+dialstr2+")
CONNECTED! ***********\n");
        }
}

exit();
----------------

Thanks!


Nicolas


On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> Hi Nicolas,
> do you have a copy of the .js code you can paste. I would guess tho, that
> ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
> false. Just a guess tho.
>
> Hangup causes can be found here:
> http://wiki.freeswitch.org/wiki/Hangup_causes
>
> <http://wiki.freeswitch.org/wiki/Hangup_causes> --matt
> hello hunter - hosted predictive dialer & voice broadcasting
> http://www.hellohunter.com
>
>
> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> I'm bridging 2 calls in a javascript file, I originate the first call and
>> then execute a bridge with an origination string for the second call. If I
>> hangup the first call while trying to make the second call, I get this on
>> the console:
>>
>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
>> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
>> sofia/external/005622170039 [KILL]
>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/005622170039 [BREAK]
>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
>> Cause: ORIGINATOR_CANCEL
>>
>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
>> NORMAL_CLEARING. And the variable_originate_disposition has a value of
>> "failure". Where can I get the detail of the call/bridge failure due to
>> 'ORIGINATOR_CANCEL' as reported through the console?
>>
>> Thanks!
>>
>> Nicolas
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jim.page at redmatter.com  Thu Aug  6 14:55:08 2009
From: jim.page at redmatter.com (Jim Page)
Date: Thu, 6 Aug 2009 22:55:08 +0100
Subject: [Freeswitch-users] CURL directory issue
In-Reply-To: <ab57a3e80908060848x64bbc808r162273eb4cbcd363@mail.gmail.com>
References: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>
	<ab57a3e80908060848x64bbc808r162273eb4cbcd363@mail.gmail.com>
Message-ID: <C45A122C-C1C0-4267-9028-C192CED52ED4@redmatter.com>

Spot on. Many thanks!
Jim

Sent from my iPhone

On 6 Aug 2009, at 18:02, "Kevin Green" <kevin at johnnyvoip.com<mailto:kevin at johnnyvoip.com>> wrote:

Try returning the full information on the register. It may be that the variables are read onto the user profile upon registration and since you are only supplying a dumbed down version for registration the variables aren't being read and cached.

Regards,
   Kevin Green

On Thu, Aug 6, 2009 at 7:24 AM, Jim Page <<mailto:jim.page at redmatter.com>jim.page at redmatter.com<mailto:jim.page at redmatter.com>> wrote:
Afternoon All

I wonder if someone (perhaps even the illustrious intralanman) could help me out with a problem I am experiencing with a CURL directory.

In the interests of understanding how the mechanism works, I am using a super-braindead php script to return info about a specific set of users. I plan to move to something more sophisticated once the proof of concept is complete, possibly based on intralanman's scripts.

The basic problem is that all works fine (boot, register, voicemail etc), except that user's <variables> seem not to be being read correctly, eg 'toll_allow' and 'user_context'. Here's a typical user XML message I am returning:

<document type="freeswitch/xml">
 <section name="directory">
  <domain name="<http://pbx.redmatter.com>pbx.redmatter.com<http://pbx.redmatter.com>">
   <params>
     <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
   </params>

   <variables>
     <variable name="record_stereo" value="true"/>
     <variable name="default_gateway" value="$${default_provider}"/>
     <variable name="default_areacode" value="$${default_areacode}"/>
     <variable name="transfer_fallback_extension" value="operator"/>
   </variables>

   <groups>
     <group name="default">
       <users>            <user id="1009" cidr="172.30.99.0/24<http://172.30.99.0/24>">
              <params>
                <param name="password" value="$${default_password}"/>
                <param name="vm-password" value="1009"/>
                <param name="vm-email-all-messages" value="true"/>
                <param name="vm-attach-file" value="true"/>
                <param name="vm-mailto" value="<mailto:1009 at redmatter.com>1009 at redmatter.com<mailto:1009 at redmatter.com>"/>
                <param name="vm-keep-local-after-email" value="true"/>
                <param name="vm_message_ext" value="mp3"/>
              </params>
              <variables>
                <variable name="toll_allow" value="domestic,international,local"/>
                <variable name="accountcode" value="1009"/>
                <variable name="user_context" value="default"/>
                <variable name="effective_caller_id_name" value="Extension 1009"/>
                <variable name="effective_caller_id_number" value="1009"/>
                <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
                <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
                <variable name="callgroup" value="techsupport"/>
              </variables>
            </user>        </users>
     </group>
   </groups>
  </domain>
 </section>
</document>

I return this kind of message in all cases except the (sip_auth_method=="REGISTER") request message where I return

<document type="freeswitch/xml">
 <section name="directory">
   <domain name="<http://pbx.redmatter.com>pbx.redmatter.com<http://pbx.redmatter.com>">
     <user id="1007">
       <params>
         <param name="password" value="1234"/>
       </params>
     </user>
   </domain>
 </section>
</document>

Also it's probably worth mentioning that I have removed all trace of xml from conf/directory and I don't believe there is a conflict happening there.

The phones register correctly. The trouble is they don't operate on the correct dialplan context (I fixed that by hardcoding the internal gateway to dialplan default), but the 'toll_allow' variable is now not working so that outbound calls fail, which is what made me think that the user variables are being ignored.

Freeswitch version is 1.0.4, built by me and running on a dell 1950 running Centos 5.3 x86_64. HTTP application is running on standard Centos 5.3 apache/php.

Any ideas gratefully and humbly received.

All the best
Jim

_______________________________________________
FreeSWITCH-users mailing list
<mailto:FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://www.freeswitch.org>http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Nick.Lemberger at lkfd.net  Thu Aug  6 16:19:03 2009
From: Nick.Lemberger at lkfd.net (Nick Lemberger)
Date: Thu, 06 Aug 2009 18:19:03 -0500
Subject: [Freeswitch-users] Lua Script Return Value & mod_xmlrpc
Message-ID: <4A7B1EC4.2C9A.00FE.0@lkfd.net>

Is it possible to have a LUA script return something to the client when accessed via the XML RPC gateway & luarun?

ie: access the url: http://FSip:8080/api/luarun?myscript.lua and have the script return a value?

-Nick



From pete at privateconnect.com  Thu Aug  6 16:35:30 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Thu, 06 Aug 2009 16:35:30 -0700
Subject: [Freeswitch-users]
	=?utf-8?q?Lua_Script_Return_Value_=26_mod=5Fxm?= =?utf-8?q?lrpc?=
Message-ID: <20090806163530.2ad02225396a31c9de30536f2e338977.1282ed1d44.wbe@email04.secureserver.net>

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From raffaele.p.guidi at gmail.com  Thu Aug  6 17:23:59 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Fri, 7 Aug 2009 02:23:59 +0200
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
	<1249555620.5449.1.camel@eli-desktop>
	<EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>
Message-ID: <b0a8792f0908061723q4fe2a24cobea858addbf4e07@mail.gmail.com>

>> 5) Seems that only EventConsumer("all") working.
EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
idea to this?

isn't it CHANNEL_HAN*G*UP? Is the G missing only in the email or in the
code, too?

On Thu, Aug 6, 2009 at 17:52, Seven Du <dujinfang at gmail.com> wrote:

> for e in (function() return con:pop(1) end) do
>
> btw, the script works.
>
> Thanks.
> On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
> > Hi
> > I dont know about events so much but I cannot see variable "e" is
> > setting
> >
> > event_name = e:getHeader("Event-Name") or ""
> > event_subclass = e:getHeader("Event-Subclass") or ""
> >
> > regurds
> > Eli
> >
> > On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
> >> ALL-
> >>
> >>
> >> I have a few questions when scripting lua. According to wiki, it is
> >> possible to run looping forever lua scripts through start-up config
> >> or luarun.
> >>
> >>
> >> 1) Will the lua script stop when unload mod_lua? I experienced core
> >> dump when unload mod_lua while there was a running lua script.
> >> Reported on jira.
> >>
> >>
> >> 2) How to stop a forever running lua script?  I stop it by
> >> listening a CUSTOM event fired elsewhere. See code below. Is there
> >> any standard way like luastop ?
> >>
> >>
> >> 3) Any way to show how many running lua scripts? luashow ?
> >>
> >>
> >> 4) It seems cannot get the lua script name in a lua script, I made
> >> a patch to jira by assign it to the argv[0].
> >>
> >>
> >> 5) Seems that only EventConsumer("all") working.
> >> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to
> >> work. Any idea to this?
> >>
> >>
> >> Thanks a lot.
> >>
> >>
> >>
> >>
> >>
> >> code example:
> >>
> >>
> >> con = freeswitch.EventConsumer("all");
> >>
> >>
> >> argv[0] = "test.lua"
> >>
> >> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]
> >> Starting =====\n");
> >>
> >> local all_events = 0
> >>
> >> for e in (function() return con:pop(1) end) do
> >>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
> >>    all_events = all_events + 1;
> >> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
> >>
> >> event_name = e:getHeader("Event-Name") or ""
> >> event_subclass = e:getHeader("Event-Subclass") or ""
> >>
> >> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
> >>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
> >> "]---Exiting------\n")
> >>  break
> >> end
> >>
> >>
> >> end
> >>
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vladrodionov at gmail.com  Thu Aug  6 17:55:55 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Thu, 6 Aug 2009 17:55:55 -0700
Subject: [Freeswitch-users] Lua on Windows and additional modules
Message-ID: <3c233920908061755h5d17aa0as3fb24743215a8298@mail.gmail.com>

Good evening,
This is newbie question.

The FreeSWITCH lua module does not support sockets and sql out of box that
is why
I just installed LuaBinaries (including socket, sql modules). My dev
environment is Win XP not Linux/Unix.

I am trying to understand what will happen when lua_module get this:

require "socket" or
require "luasql.mysql"
?

How does lua_module look up additional lua modules on Windows platform?

Do I have to set some env variables?

TIA
-Vladimir Rodionov
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From dujinfang at gmail.com  Thu Aug  6 20:45:16 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 7 Aug 2009 11:45:16 +0800
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <b0a8792f0908061723q4fe2a24cobea858addbf4e07@mail.gmail.com>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
	<1249555620.5449.1.camel@eli-desktop>
	<EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>
	<b0a8792f0908061723q4fe2a24cobea858addbf4e07@mail.gmail.com>
Message-ID: <23f91030908062045s6ebc35efne6f5ce999f085506@mail.gmail.com>

Sorry it's a typo. I read the code, it works not like in event socket. So,
only works with one event.
either
EventConsumer("all")
or
EventConsumer("CUSTOM", "lua::stop");


Thank you.

2009/8/7 Raffaele P. Guidi <raffaele.p.guidi at gmail.com>

> >> 5) Seems that only EventConsumer("all") working.
> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
> idea to this?
>
> isn't it CHANNEL_HAN*G*UP? Is the G missing only in the email or in the
> code, too?
>
> On Thu, Aug 6, 2009 at 17:52, Seven Du <dujinfang at gmail.com> wrote:
>
>> for e in (function() return con:pop(1) end) do
>>
>> btw, the script works.
>>
>> Thanks.
>> On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
>> > Hi
>> > I dont know about events so much but I cannot see variable "e" is
>> > setting
>> >
>> > event_name = e:getHeader("Event-Name") or ""
>> > event_subclass = e:getHeader("Event-Subclass") or ""
>> >
>> > regurds
>> > Eli
>> >
>> > On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
>> >> ALL-
>> >>
>> >>
>> >> I have a few questions when scripting lua. According to wiki, it is
>> >> possible to run looping forever lua scripts through start-up config
>> >> or luarun.
>> >>
>> >>
>> >> 1) Will the lua script stop when unload mod_lua? I experienced core
>> >> dump when unload mod_lua while there was a running lua script.
>> >> Reported on jira.
>> >>
>> >>
>> >> 2) How to stop a forever running lua script?  I stop it by
>> >> listening a CUSTOM event fired elsewhere. See code below. Is there
>> >> any standard way like luastop ?
>> >>
>> >>
>> >> 3) Any way to show how many running lua scripts? luashow ?
>> >>
>> >>
>> >> 4) It seems cannot get the lua script name in a lua script, I made
>> >> a patch to jira by assign it to the argv[0].
>> >>
>> >>
>> >> 5) Seems that only EventConsumer("all") working.
>> >> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to
>> >> work. Any idea to this?
>> >>
>> >>
>> >> Thanks a lot.
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> code example:
>> >>
>> >>
>> >> con = freeswitch.EventConsumer("all");
>> >>
>> >>
>> >> argv[0] = "test.lua"
>> >>
>> >> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]
>> >> Starting =====\n");
>> >>
>> >> local all_events = 0
>> >>
>> >> for e in (function() return con:pop(1) end) do
>> >>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>> >>    all_events = all_events + 1;
>> >> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
>> >>
>> >> event_name = e:getHeader("Event-Name") or ""
>> >> event_subclass = e:getHeader("Event-Subclass") or ""
>> >>
>> >> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>> >>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
>> >> "]---Exiting------\n")
>> >>  break
>> >> end
>> >>
>> >>
>> >> end
>> >>
>> >>
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Thu Aug  6 21:07:00 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 7 Aug 2009 11:07:00 +0700
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 Release Announcement
In-Reply-To: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com>
References: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com>
Message-ID: <8ccbff060908062107p1f21fd8dp8c631d7a627520f8@mail.gmail.com>

Good News..

2009/8/7 Michael Collins <msc at freeswitch.org>:
> We are happy to announce the official release of FreeSWITCH 1.0.4! Please
> visit this link to Digg and read the story, and then spread the word!
>
> Thanks for being such a great community!
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From pete at privateconnect.com  Thu Aug  6 21:59:34 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Thu, 06 Aug 2009 21:59:34 -0700
Subject: [Freeswitch-users] Lua on Windows and additional modules
Message-ID: <20090806215934.2ad02225396a31c9de30536f2e338977.122e91f14a.wbe@email04.secureserver.net>

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From velu.technical at gmail.com  Thu Aug  6 22:06:21 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Fri, 7 Aug 2009 10:36:21 +0530
Subject: [Freeswitch-users] Fwd: execute function in ESL.pm module is
	not working
In-Reply-To: <87f2f3b90908052254l3ecc7fa0ybc92d87c587a9b0d@mail.gmail.com>
References: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
	<1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>
	<87f2f3b90908052254l3ecc7fa0ybc92d87c587a9b0d@mail.gmail.com>
Message-ID: <1452e2980908062206s3b33f867if569a10fff078673@mail.gmail.com>

Dear Expert,
   Thanks for you reply....

My Perl Script is,
use strict;
use warnings;

#---------------------------------------------------------------------------
# Event socket library.
# Socket programming
# printing the data structures
# Using posix parametered functions.
#---------------------------------------------------------------------------
use lib('/root/freeswitch-1.0.3/libs/esl/perl/');
require ESL;
use IO::Socket::INET;
use Data::Dumper qw(Dumper);
use POSIX;
use Config::IniFiles;


# Global variables to store the socket connection and eneterd DTM digits.
my ($conn,$digit);
$digit='';

#Registering the ALARM signal.
$SIG{ALRM}=\&sub_alr;

# When alarm signal occurs call the play_digit function
sub sub_alr {
        print "IN Sigalarm---\n";
        &play_digit;
        return ;
}       # ----------  end of subroutine sub_alr  ----------



# Play the voice files for menu.
sub play(){
                $conn->execute("playback","ivr/ivr-please.wav");
                $conn->execute("playback","ivr/ivr-enter_ext.wav");
}


sub play_digit {
        print "In Play Digit....\n";
        my      ( $par1 )       = $digit; #$digit is global variable

        print "Eneterd Digits=",$digit,"\n";
        ################################################################
        # Here what is my problem the execute function is not working  #
        ################################################################
        $conn->execute("phrase", "spell,$par1");
        return ;
}       # ----------  end of subroutine play_digit  ----------


#---------------------------------------------------------------------------
# IP address and port of the server.
# Sound path file.
#---------------------------------------------------------------------------
my $ip = "192.168.1.222"; my $port = '5057';
my $sound_path = "/usr/local/freeswitch/sounds/en/us/callie/";


# Creating a socket
my $sock = new IO::Socket::INET (
        LocalHost => $ip,
        LocalPort => $port,
        Proto     => 'tcp',
        Listen    => 1,
        Reuse     => 1
);
# Checking the error.
die "Cannot create a socket:$!\n" unless $sock;


for(;;){

        my $new_socket = $sock->accept();
        print "Current Process Id:".POSIX::getpid()."\n";
        my $pid = fork();

        if($pid){
                close($new_socket);
                next;
        }

        print "Child Process Id:".POSIX::getpid()."\n";

        my $fd = fileno($new_socket);
        print "File Number:$fd\n";

        # Create a conenction with Event socket library.
        $conn = new ESL::ESLconnection($fd);

        # Getting the connection informations and values of the variables.
        my $info = $conn->getInfo();

        # Getting the caller id and print the statement.
        my $caller_id =$info->getHeader("caller-caller-id-number");
        printf  "Connected from %s\n", $caller_id;

        # Receive the events from only in this switch.
        $conn->sendRecv("myevents");

        # Answer the call.
        $conn->execute("answer");

        # playback the welcome message.
        $conn->setEventLock("true");

$conn->execute("playback",$sound_path."ivr/ivr-welcome_to_freeswitch.wav");
        $conn->execute("sleep", "1000");

        &play;


        alarm(10);
        while($conn->connected()){

                # Receive the event
                my $event = $conn->recvEvent();

                # Check the event is received
                if($event){
                        # Get the event name  and print it.
                        my $name = $event->getHeader("event-name");
                        print "EVENT:[$name]\n";

                        # If the event name is DTMF then print the  enterted
digit.
                        if($name eq 'DTMF'){
                                my $digi = $event->getHeader("dtmf-digit");
                                # Here concatenate the eneterd digits
                                $digit.=$digi;
                        }
                }
        }
        # Kill the child process.
        print "Disconnected:$caller_id\n"; kill 9,POSIX::getpid();
}


My dial plan is,
<!-- Testing IVR -->
    <extension name="Test">
      <condition field="destination_number" expression="^(200)$">
            <action application="socket" data="192.168.1.222:5057 async
full"/>
      </condition>
    </extension>

The output of the Script is,

Current Process Id:2906
Child Process Id:2908
File Number:4
Connected from 1000
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_ANSWER]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[DTMF]
EVENT:[DTMF]
EVENT:[DTMF]
EVENT:[DTMF]
IN Sigalarm---
In Play Digit....
Eneterd Digits=7485
Disconnected:1000

When alarm signal generated, it prints digits but it won't execute the
"execute" function..

Please  any one give suggestions where I made wrong...

Thanks...

Regards,
Velusamy.


On Thu, Aug 6, 2009 at 11:24 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 11:38 PM, velusamy velu <velu.technical at gmail.com>wrote:
>
>> Please any one help for this problem..
>>
>>
> Sorry for the delay but many of the FreeSWITCH experts are at ClueCon right
> now so we'll ask for your patience... in the meantime could you pastebin
> your script and your dialplan entry so that we can take a look at them?
>
> Thanks,
> MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ryder86 at googlemail.com  Fri Aug  7 00:36:16 2009
From: ryder86 at googlemail.com (Artem Vasiliev)
Date: Fri, 7 Aug 2009 11:36:16 +0400
Subject: [Freeswitch-users] Softphone control
Message-ID: <cb6be45b0908070036x5063615ev93e2526427905a23@mail.gmail.com>

Hi

I have FreeSwitch and external application, which communicates to it via
event socket - listens for events for certain number and gives some
commands.
Is it possible for this application to control client softphones, for
example, make them answer or hold, using the event socket or other
FreeSwitch capabilities?
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From dujinfang at gmail.com  Fri Aug  7 01:10:17 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 7 Aug 2009 16:10:17 +0800
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <cb6be45b0908070036x5063615ev93e2526427905a23@mail.gmail.com>
References: <cb6be45b0908070036x5063615ev93e2526427905a23@mail.gmail.com>
Message-ID: <23f91030908070110j199fc1fcpcce4685318a1e8c9@mail.gmail.com>

You can run FreeSWITCH as a softphone and control it.
http://wiki.freeswitch.org/wiki/Freeswitch_softphone

2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>

> Hi
>
> I have FreeSwitch and external application, which communicates to it via
> event socket - listens for events for certain number and gives some
> commands.
> Is it possible for this application to control client softphones, for
> example, make them answer or hold, using the event socket or other
> FreeSwitch capabilities?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ryder86 at googlemail.com  Fri Aug  7 04:02:04 2009
From: ryder86 at googlemail.com (Artem Vasiliev)
Date: Fri, 7 Aug 2009 15:02:04 +0400
Subject: [Freeswitch-users] Softphone control
Message-ID: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>

No, I don't want to make softphone from FreeSwitch

I have FS and several users with eyeBeam softphones. I need to control
those eyeBeams

>You can run FreeSWITCH as a softphone and control it.
>http://wiki.freeswitch.org/wiki/Freeswitch_softphone

>2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>

>> Hi
>>
>> I have FreeSwitch and external application, which communicates to it via
>> event socket - listens for events for certain number and gives some
>> commands.
>> Is it possible for this application to control client softphones, for
>> example, make them answer or hold, using the event socket or other
>> FreeSwitch capabilities?
>>


From merul at mac.com  Fri Aug  7 04:17:12 2009
From: merul at mac.com (Merul Patel)
Date: Fri, 07 Aug 2009 12:17:12 +0100
Subject: [Freeswitch-users] /etc/openzap/tones.conf for UK
Message-ID: <599DBC67-02FE-4F9E-9F87-6D1749B81B11@mac.com>

Where can I find a sample tones.conf file for the UK? Am trying to  
configure a USBFXO device for outbound calls.

Thanks in advance,

Merul


From kevin at johnnyvoip.com  Fri Aug  7 08:00:39 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 7 Aug 2009 11:00:39 -0400
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
Message-ID: <ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>

>From what I am aware you can't use FreeSWITCH to control a softphone
directly though you can make it do things that will have a similar end
result. You could set eyeBeam to auto-answer calls if you want them to
answer right away or orginiate a call that is auto-answered but not bridge
the call until a user on the eyeBeam presses a digit or a socket control
tells it to connect the two ends. You can also use FreeSWITCH to place the
line on hold using event sockets, this will place it on hold in the server
and not directly like placing it on hold in eyeBeam (i.e. the hold button in
eyeBeam likely wont show it as being on hold).

Beyond that if you want to directly control the clients you would need to
look at getting an API access into the eyeBeam client.

I hope this will help.

Regards,
   Kevin Green


On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>wrote:

> No, I don't want to make softphone from FreeSwitch
>
> I have FS and several users with eyeBeam softphones. I need to control
> those eyeBeams
>
> >You can run FreeSWITCH as a softphone and control it.
> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>
> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>
> >> Hi
> >>
> >> I have FreeSwitch and external application, which communicates to it via
> >> event socket - listens for events for certain number and gives some
> >> commands.
> >> Is it possible for this application to control client softphones, for
> >> example, make them answer or hold, using the event socket or other
> >> FreeSwitch capabilities?
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From asannucci at gmail.com  Fri Aug  7 08:21:51 2009
From: asannucci at gmail.com (bakko)
Date: Fri, 7 Aug 2009 17:21:51 +0200
Subject: [Freeswitch-users] Spanish Prompts
Message-ID: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>

I'd like to begin record spanish prompts for FS.

Do you know any software/hardware to make it?

Thank you

BR



From nicolas at medularis.com  Fri Aug  7 09:43:29 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 12:43:29 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
	<1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
Message-ID: <1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>

I changed the script to set hangup_after_bridge to false, but still the same
thing happens, I get this on the console:

2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup
sofia/external/00569xxxxxxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/00569xxxxxxx [KILL]
2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/00569xxxxxxx [BREAK]
2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398
(sofia/external/00569xxxxxxx) Running State Change CS_HANGUP
2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL
2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *********** Leg2:
NORMAL_CLEARING ***********


The second to last line comes from the script, and prints the hangup_cause
of he session, instead of getting ORIGINATOR_CANCEL, I'm getting
NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?


Thanks!

Nicolas

On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi Matt,
>
> Actually I'm explicitly setting hangup_after_bridge to true, think setting
> it to false would help? I'm going to try that.
>
> Here's the JS code:
> (Note: session.getVariable() doesn't work, FS complains saying it is not a
> function, also tried self.session.getVariable() - that's what the wiki says
> - and FS complains that self does not exist)
>
> ----------------
> var uuid = argv[0]; // Call identifier
> var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
> var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
> var greeting_snd = "/var/audio/alert.wav";
>
> console_log("notice", "*********** STARTING C2C Call ***********\n");
> timeout = 30;
>
> console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");
>
> //var stUsRing = session.getVariable("us-ring");  // This doesn't work,
> self.session.getVariable doesn't work either
> var stUsRing = "%(2000,4000,440,480)";
>
> // Create new_session
> new_session = new Session(originate_str1);
> console_log("notice", "*********** Leg1: " + new_session.cause + "
> ***********\n");
>
> if (new_session.ready()) {
>         // log to the console
>         console_log("notice", "*********** Leg1 ("+dialstr1+") CONNECTED!
> ***********\n");
>         console_log("notice", "*********** Playing greeting sound:
> "+greeting_snd+" ***********\n");
>
>         new_session.execute("sleep", 100);
>         new_session.execute("playback", greeting_snd);
>
>         // Originate second call and bridge
>     originate_str2 =
> "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
>
>         // Create new_session
>         new_session.execute("bridge", originate_str2);
>         console_log("notice", "*********** Leg2: " + new_session.cause + "
> ***********\n");
>
>         if (new_session.ready()) {
>                 console_log("notice", "*********** Leg2 ("+dialstr2+")
> CONNECTED! ***********\n");
>         }
> }
>
> exit();
> ----------------
>
> Thanks!
>
>
> Nicolas
>
>
>
> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> Hi Nicolas,
>> do you have a copy of the .js code you can paste. I would guess tho, that
>> ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
>> false. Just a guess tho.
>>
>> Hangup causes can be found here:
>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>
>> <http://wiki.freeswitch.org/wiki/Hangup_causes> --matt
>> hello hunter - hosted predictive dialer & voice broadcasting
>> http://www.hellohunter.com
>>
>>
>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> I'm bridging 2 calls in a javascript file, I originate the first call and
>>> then execute a bridge with an origination string for the second call. If I
>>> hangup the first call while trying to make the second call, I get this on
>>> the console:
>>>
>>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
>>> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
>>> sofia/external/005622170039 [KILL]
>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/005622170039 [BREAK]
>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
>>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
>>> Cause: ORIGINATOR_CANCEL
>>>
>>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
>>> NORMAL_CLEARING. And the variable_originate_disposition has a value of
>>> "failure". Where can I get the detail of the call/bridge failure due to
>>> 'ORIGINATOR_CANCEL' as reported through the console?
>>>
>>> Thanks!
>>>
>>> Nicolas
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From pjintheusa at gmail.com  Fri Aug  7 10:28:23 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 7 Aug 2009 13:28:23 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
	<1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
	<1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>
Message-ID: <367751820908071028q7075b710hb0d6eed8c1d4dc54@mail.gmail.com>

What does

bridge_hangup_cause

give you?

On Fri, Aug 7, 2009 at 12:43 PM, Nicolas Brenner <nicolas at medularis.com> wrote:
>
> I changed the script to set hangup_after_bridge to false, but still the same thing happens, I get this on the console:
>
> 2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup sofia/external/00569xxxxxxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal sofia/external/00569xxxxxxx [KILL]
> 2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal sofia/external/00569xxxxxxx [BREAK]
> 2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal sofia/external/005622170039 [KILL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal sofia/external/005622170039 [BREAK]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398 (sofia/external/00569xxxxxxx) Running State Change CS_HANGUP
> 2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.? Cause: ORIGINATOR_CANCEL
> 2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *********** Leg2: NORMAL_CLEARING ***********
>
>
> The second to last line comes from the script, and prints the hangup_cause of he session, instead of getting ORIGINATOR_CANCEL, I'm getting NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?
>
>
> Thanks!
>
> Nicolas
>
> On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner <nicolas at medularis.com> wrote:
>>
>> Hi Matt,
>>
>> Actually I'm explicitly setting hangup_after_bridge to true, think setting it to false would help? I'm going to try that.
>>
>> Here's the JS code:
>> (Note: session.getVariable() doesn't work, FS complains saying it is not a function, also tried self.session.getVariable() - that's what the wiki says - and FS complains that self does not exist)
>>
>> ----------------
>> var uuid = argv[0]; // Call identifier
>> var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
>> var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
>> var greeting_snd = "/var/audio/alert.wav";
>>
>> console_log("notice", "*********** STARTING C2C Call ***********\n");
>> timeout = 30;
>>
>> console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");
>>
>> //var stUsRing = session.getVariable("us-ring");? // This doesn't work, self.session.getVariable doesn't work either
>> var stUsRing = "%(2000,4000,440,480)";
>>
>> // Create new_session
>> new_session = new Session(originate_str1);
>> console_log("notice", "*********** Leg1: " + new_session.cause + " ***********\n");
>>
>> if (new_session.ready()) {
>> ??????? // log to the console
>> ??????? console_log("notice", "*********** Leg1 ("+dialstr1+") CONNECTED! ***********\n");
>> ??????? console_log("notice", "*********** Playing greeting sound: "+greeting_snd+" ***********\n");
>>
>> ??????? new_session.execute("sleep", 100);
>> ??????? new_session.execute("playback", greeting_snd);
>>
>> ??????? // Originate second call and bridge
>> ??? originate_str2 = "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
>>
>> ??????? // Create new_session
>> ??????? new_session.execute("bridge", originate_str2);
>> ??????? console_log("notice", "*********** Leg2: " + new_session.cause + " ***********\n");
>>
>> ??????? if (new_session.ready()) {
>> ??????????????? console_log("notice", "*********** Leg2 ("+dialstr2+") CONNECTED! ***********\n");
>> ??????? }
>> }
>>
>> exit();
>> ----------------
>>
>> Thanks!
>>
>>
>> Nicolas
>>
>>
>> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>>>
>>> Hi Nicolas,
>>> do you have a copy of the .js code you can paste. I would guess tho, that ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to false. Just a guess tho.
>>> Hangup causes can be found here:
>>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>> --matt
>>> hello hunter - hosted predictive dialer & voice broadcasting
>>> http://www.hellohunter.com
>>>
>>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com> wrote:
>>>>
>>>> I'm bridging 2 calls in a javascript file, I originate the first call and then execute a bridge with an origination string for the second call. If I hangup the first call while trying to make the second call, I get this on the console:
>>>>
>>>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal sofia/external/005622170039 [KILL]
>>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal sofia/external/005622170039 [BREAK]
>>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>>>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.? Cause: ORIGINATOR_CANCEL
>>>>
>>>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see NORMAL_CLEARING. And the variable_originate_disposition has a value of "failure". Where can I get the detail of the call/bridge failure due to 'ORIGINATOR_CANCEL' as reported through the console?
>>>>
>>>> Thanks!
>>>>
>>>> Nicolas
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From raffaele.p.guidi at gmail.com  Fri Aug  7 11:50:57 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Fri, 7 Aug 2009 20:50:57 +0200
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
	<ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
Message-ID: <b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>

Maybe Artem is interested in CTI (computer telephony integration) -
click2dial, opening a url (or statrting a program) on incoming call...?

On Fri, Aug 7, 2009 at 17:00, Kevin Green <kevin at johnnyvoip.com> wrote:

> From what I am aware you can't use FreeSWITCH to control a softphone
> directly though you can make it do things that will have a similar end
> result. You could set eyeBeam to auto-answer calls if you want them to
> answer right away or orginiate a call that is auto-answered but not bridge
> the call until a user on the eyeBeam presses a digit or a socket control
> tells it to connect the two ends. You can also use FreeSWITCH to place the
> line on hold using event sockets, this will place it on hold in the server
> and not directly like placing it on hold in eyeBeam (i.e. the hold button in
> eyeBeam likely wont show it as being on hold).
>
> Beyond that if you want to directly control the clients you would need to
> look at getting an API access into the eyeBeam client.
>
> I hope this will help.
>
> Regards,
>    Kevin Green
>
>
>
> On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>wrote:
>
>> No, I don't want to make softphone from FreeSwitch
>>
>> I have FS and several users with eyeBeam softphones. I need to control
>> those eyeBeams
>>
>> >You can run FreeSWITCH as a softphone and control it.
>> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>
>> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>>
>> >> Hi
>> >>
>> >> I have FreeSwitch and external application, which communicates to it
>> via
>> >> event socket - listens for events for certain number and gives some
>> >> commands.
>> >> Is it possible for this application to control client softphones, for
>> >> example, make them answer or hold, using the event socket or other
>> >> FreeSwitch capabilities?
>> >>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lfurrea at gmail.com  Fri Aug  7 11:57:55 2009
From: lfurrea at gmail.com (Luis F Urrea)
Date: Fri, 7 Aug 2009 12:57:55 -0600
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
Message-ID: <f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>

On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
edition .

Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for FS
playback.

Here's a guide that has been put together for reference on what to record.

http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml

Regards,



On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:

> I'd like to begin record spanish prompts for FS.
>
> Do you know any software/hardware to make it?
>
> Thank you
>
> BR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Fri Aug  7 12:10:35 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 7 Aug 2009 15:10:35 -0400
Subject: [Freeswitch-users] Calling multiple destinations with fail over
Message-ID: <367751820908071210y233741fatad5cd451dd64df7@mail.gmail.com>

Hi there,

I am trying to implement a scenario where I can terminate calls to
multiple destinations AND have termination carrier fail over.
Currently I can see how to do one or the other. But not both.

Multiple destinations is easy:

 <action application="bridge"
data="[origination_caller_id_number=1234]sofia/SIP_PROVIDER_1/6095551234,[origination_caller_id_number=55555]sofia/SIP_PROVIDER_1/7325551234"/>

Failover appears to use the same mechanism:

<action application="bridge"
data="sofia/gateway/primary/blah|sofia/gateway/secondary/blah"/>

I can not get my head around how use these together, such that:


- 6095551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 6095551234  through SIP_PROVIDER_2
- Called party does not answer
7325551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2
- Called party answers


This must be a fairly comment requirement so any ideas on what I might
be missing would be very welcome.

Thanks


Phillip Jones


From jmesquita at gmail.com  Fri Aug  7 13:50:32 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 7 Aug 2009 17:50:32 -0300
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
	<ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
	<b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>
Message-ID: <5a8712120908071350v2eac4613j7fd53e5158680742@mail.gmail.com>

Stay tuned on fsgui. It will get there really soon.

jmesquita

On Fri, Aug 7, 2009 at 3:50 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Maybe Artem is interested in CTI (computer telephony integration) -
> click2dial, opening a url (or statrting a program) on incoming call...?
>
>
> On Fri, Aug 7, 2009 at 17:00, Kevin Green <kevin at johnnyvoip.com> wrote:
>
>> From what I am aware you can't use FreeSWITCH to control a softphone
>> directly though you can make it do things that will have a similar end
>> result. You could set eyeBeam to auto-answer calls if you want them to
>> answer right away or orginiate a call that is auto-answered but not bridge
>> the call until a user on the eyeBeam presses a digit or a socket control
>> tells it to connect the two ends. You can also use FreeSWITCH to place the
>> line on hold using event sockets, this will place it on hold in the server
>> and not directly like placing it on hold in eyeBeam (i.e. the hold button in
>> eyeBeam likely wont show it as being on hold).
>>
>> Beyond that if you want to directly control the clients you would need to
>> look at getting an API access into the eyeBeam client.
>>
>> I hope this will help.
>>
>> Regards,
>>     Kevin Green
>>
>>
>>
>> On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>wrote:
>>
>>> No, I don't want to make softphone from FreeSwitch
>>>
>>> I have FS and several users with eyeBeam softphones. I need to control
>>> those eyeBeams
>>>
>>> >You can run FreeSWITCH as a softphone and control it.
>>> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>>
>>> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>>>
>>> >> Hi
>>> >>
>>> >> I have FreeSwitch and external application, which communicates to it
>>> via
>>> >> event socket - listens for events for certain number and gives some
>>> >> commands.
>>> >> Is it possible for this application to control client softphones, for
>>> >> example, make them answer or hold, using the event socket or other
>>> >> FreeSwitch capabilities?
>>> >>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Fri Aug  7 14:20:29 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 17:20:29 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <367751820908071028q7075b710hb0d6eed8c1d4dc54@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
	<1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
	<1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>
	<367751820908071028q7075b710hb0d6eed8c1d4dc54@mail.gmail.com>
Message-ID: <1b46b4e80908071420v42c3f73wfeddacb8af3d7067@mail.gmail.com>

That variable is not available, it is not included with the
CHANNEL_HANGUP_COMPLETE event info. However I discovered that when the
bridge does not work, there are two CHANNEL_HANGUP_COMPLETE events, one for
each leg, nevertheless for some reason the daemon I have watching the events
misses the second leg event, so I was only seeing the result of the first
leg hangup, which is NORMAL_CLEARING, and the second event's hangup_cause is
ORIGINATOR_CANCEL. I don't know why my daemon is missing the event though.
I'll have to dig into this further.


On Fri, Aug 7, 2009 at 1:28 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> What does
>
> bridge_hangup_cause
>
> give you?
>
> On Fri, Aug 7, 2009 at 12:43 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> >
> > I changed the script to set hangup_after_bridge to false, but still the
> same thing happens, I get this on the console:
> >
> > 2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup
> sofia/external/00569xxxxxxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> > 2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/00569xxxxxxx [KILL]
> > 2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/00569xxxxxxx [BREAK]
> > 2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/005622170039 [KILL]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/005622170039 [BREAK]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate
> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/00569xxxxxxx) Running State Change CS_HANGUP
> > 2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.
> Cause: ORIGINATOR_CANCEL
> > 2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *********** Leg2:
> NORMAL_CLEARING ***********
> >
> >
> > The second to last line comes from the script, and prints the
> hangup_cause of he session, instead of getting ORIGINATOR_CANCEL, I'm
> getting NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?
> >
> >
> > Thanks!
> >
> > Nicolas
> >
> > On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> >>
> >> Hi Matt,
> >>
> >> Actually I'm explicitly setting hangup_after_bridge to true, think
> setting it to false would help? I'm going to try that.
> >>
> >> Here's the JS code:
> >> (Note: session.getVariable() doesn't work, FS complains saying it is not
> a function, also tried self.session.getVariable() - that's what the wiki
> says - and FS complains that self does not exist)
> >>
> >> ----------------
> >> var uuid = argv[0]; // Call identifier
> >> var dialstr1 = argv[1]; // Dial string obtained from previous call to
> LCR
> >> var dialstr2 = argv[2]; // Dial string obtained from previous call to
> LCR
> >> var greeting_snd = "/var/audio/alert.wav";
> >>
> >> console_log("notice", "*********** STARTING C2C Call ***********\n");
> >> timeout = 30;
> >>
> >> console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");
> >>
> >> //var stUsRing = session.getVariable("us-ring");  // This doesn't work,
> self.session.getVariable doesn't work either
> >> var stUsRing = "%(2000,4000,440,480)";
> >>
> >> // Create new_session
> >> new_session = new Session(originate_str1);
> >> console_log("notice", "*********** Leg1: " + new_session.cause + "
> ***********\n");
> >>
> >> if (new_session.ready()) {
> >>         // log to the console
> >>         console_log("notice", "*********** Leg1 ("+dialstr1+")
> CONNECTED! ***********\n");
> >>         console_log("notice", "*********** Playing greeting sound:
> "+greeting_snd+" ***********\n");
> >>
> >>         new_session.execute("sleep", 100);
> >>         new_session.execute("playback", greeting_snd);
> >>
> >>         // Originate second call and bridge
> >>     originate_str2 =
> "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
> >>
> >>         // Create new_session
> >>         new_session.execute("bridge", originate_str2);
> >>         console_log("notice", "*********** Leg2: " + new_session.cause +
> " ***********\n");
> >>
> >>         if (new_session.ready()) {
> >>                 console_log("notice", "*********** Leg2 ("+dialstr2+")
> CONNECTED! ***********\n");
> >>         }
> >> }
> >>
> >> exit();
> >> ----------------
> >>
> >> Thanks!
> >>
> >>
> >> Nicolas
> >>
> >>
> >> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com>
> wrote:
> >>>
> >>> Hi Nicolas,
> >>> do you have a copy of the .js code you can paste. I would guess tho,
> that ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge
> to false. Just a guess tho.
> >>> Hangup causes can be found here:
> >>> http://wiki.freeswitch.org/wiki/Hangup_causes
> >>> --matt
> >>> hello hunter - hosted predictive dialer & voice broadcasting
> >>> http://www.hellohunter.com
> >>>
> >>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> >>>>
> >>>> I'm bridging 2 calls in a javascript file, I originate the first call
> and then execute a bridge with an origination string for the second call. If
> I hangup the first call while trying to make the second call, I get this on
> the console:
> >>>>
> >>>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> >>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/005622170039 [KILL]
> >>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/005622170039 [BREAK]
> >>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134
> Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> >>>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
> Cause: ORIGINATOR_CANCEL
> >>>>
> >>>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I
> see NORMAL_CLEARING. And the variable_originate_disposition has a value of
> "failure". Where can I get the detail of the call/bridge failure due to
> 'ORIGINATOR_CANCEL' as reported through the console?
> >>>>
> >>>> Thanks!
> >>>>
> >>>> Nicolas
> >>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nicolas at medularis.com  Fri Aug  7 15:10:25 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 18:10:25 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
Message-ID: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>

Hi,

I'm trying to get started with the ESL using PHP. I compiled the ESL, then
phpmod according to the wiki instructions, but then when I try the examples
in the libs/esl/php dir, they fail saying:

PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
/usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132

Checking ESL.php on line 132, I see there are several different declarations
for the function __construct() with different parameters each, which makes
sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
higher than that or something? What could be the problem?

Thanks!

Nicolas
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From dave at 3c.co.uk  Fri Aug  7 15:54:08 2009
From: dave at 3c.co.uk (David Knell)
Date: Fri, 07 Aug 2009 17:54:08 -0500
Subject: [Freeswitch-users] Cluecon 2009
Message-ID: <1249685648.16901.34.camel@dk-d820>

Just a quick note to say thanks to Cluecon's organisers for putting
together such a useful, informative and packed three days.  I've come
away with a head full of ideas, a bunch of new contacts and a collection
of things to do; I'd thoroughly recommend that anyone interested in IP
telephony blocks out the first week of August 2010, right now..!

Cheers --

Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From gcd at i.ph  Fri Aug  7 16:44:23 2009
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 8 Aug 2009 07:44:23 +0800
Subject: [Freeswitch-users] /etc/openzap/tones.conf for UK
In-Reply-To: <599DBC67-02FE-4F9E-9F87-6D1749B81B11@mac.com>
References: <599DBC67-02FE-4F9E-9F87-6D1749B81B11@mac.com>
Message-ID: <7d0bfd8c0908071644n2933a91y996745e9f894cf90@mail.gmail.com>

you can create your tones.conf using call progress tones  found at
http://www.3amsystems.com/wireline/tone-search.htm


On Fri, Aug 7, 2009 at 7:17 PM, Merul Patel <merul at mac.com> wrote:

> Where can I find a sample tones.conf file for the UK? Am trying to
> configure a USBFXO device for outbound calls.
>
> Thanks in advance,
>
> Merul
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Fri Aug  7 16:44:41 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 7 Aug 2009 19:44:41 -0400
Subject: [Freeswitch-users] State of originated call
Message-ID: <c55d46950908071644m7604eb57jf5018c2b378d3255@mail.gmail.com>

Hi,

using javascript, i do originate the call this way:

Session s= new Session(originateStr);

>From this point, is it possible to know what states the call is going
through? In a previous message it was suggested that
variable_originate_disposition would give me the response code. Now, how to
i use this in practice in a script?
How do i for instance retrieve a 180 response code when rining is hapening
on the remote end?

Thanks,
Max.
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From nicolas at medularis.com  Fri Aug  7 17:23:51 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 20:23:51 -0400
Subject: [Freeswitch-users] Best practices / tips for Event socket daemon
Message-ID: <1b46b4e80908071723m18bf3bcnb2505f0f760b907c@mail.gmail.com>

Hi, I built an event socket daemon that waits for certain events, when it
receives those events, it does some processing and keeps waiting for more
events. The daemon is written on PHP and uses a slightly modified version of
fs_sock.php (from contrib/intralanman/PHP/fs_sock/).

What I am doing / what I want to do: I am generating calls and bridging them
using a JS script. Then the daemon logs the info about the calls and keeps
track of their status in a database.

The problem is: the daemon is apparently missing out on some events, and I
think it is because of the processing/updating on the DB it has to do each
time it "catches" an event on the socket.

My question is: which language would you recommend for the task, and how
would you go about handling events? Should the dameon fork a process for
each event it receives so that it doesn't miss any events? should there be
more than one daemon? ... any tips and recommendations are welcome.

Thanks!

Nicolas
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From rehan at supertec.com  Fri Aug  7 19:21:36 2009
From: rehan at supertec.com (Rehan Ahmed Allahwala)
Date: Sat, 8 Aug 2009 10:21:36 +0800
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <1249685648.16901.34.camel@dk-d820>
References: <1249685648.16901.34.camel@dk-d820>
Message-ID: <865f01c80908071921q13b4f98bxea7426b720a69e83@mail.gmail.com>

Hi all, hope u got ur hands on our mobile pouches from didx.net and
got ur snaps with suzanne for our blogs and new upcoming magazine

Thanks ,

Rehan

On 8/8/09, David Knell <dave at 3c.co.uk> wrote:
> Just a quick note to say thanks to Cluecon's organisers for putting
> together such a useful, informative and packed three days.  I've come
> away with a head full of ideas, a bunch of new contacts and a collection
> of things to do; I'd thoroughly recommend that anyone interested in IP
> telephony blocks out the first week of August 2010, right now..!
>
> Cheers --
>
> Dave
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From vladrodionov at gmail.com  Fri Aug  7 19:42:38 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Fri, 7 Aug 2009 19:42:38 -0700
Subject: [Freeswitch-users] Best practices / tips for Event socket daemon
In-Reply-To: <1b46b4e80908071723m18bf3bcnb2505f0f760b907c@mail.gmail.com>
References: <1b46b4e80908071723m18bf3bcnb2505f0f760b907c@mail.gmail.com>
Message-ID: <3c233920908071942g522df080g63b2abc31efd6fcc@mail.gmail.com>

Forking process on every incoming event is terrible idea IMO. Threads are
more lightweight than processes.
Can you use threads in PHP? I am not familiar with PHP (Java developer
myself).  I can explain how I would implement it in Java.

There is one SocketReader thread and several Worker threads in a thread
pool.

"SocketReader" thread - reads data (events) from socket. When event arrives
SocketReader checks thread pool, get one Worker (if any) and makes it to
process event. If there no available Workers in a pool then event goes
directly to a EventQueue. When Worker finishes it checks EventQueue and if
there are no events in a queue Worker goes back to thread pool, otherwise it
process event from queue.

-Vladimir Rodionov



On Fri, Aug 7, 2009 at 5:23 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi, I built an event socket daemon that waits for certain events, when it
> receives those events, it does some processing and keeps waiting for more
> events. The daemon is written on PHP and uses a slightly modified version of
> fs_sock.php (from contrib/intralanman/PHP/fs_sock/).
>
> What I am doing / what I want to do: I am generating calls and bridging
> them using a JS script. Then the daemon logs the info about the calls and
> keeps track of their status in a database.
>
> The problem is: the daemon is apparently missing out on some events, and I
> think it is because of the processing/updating on the DB it has to do each
> time it "catches" an event on the socket.
>
> My question is: which language would you recommend for the task, and how
> would you go about handling events? Should the dameon fork a process for
> each event it receives so that it doesn't miss any events? should there be
> more than one daemon? ... any tips and recommendations are welcome.
>
> Thanks!
>
> Nicolas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From andrew at hijacked.us  Fri Aug  7 19:56:14 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 7 Aug 2009 22:56:14 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
Message-ID: <20090808025613.GA19871@hijacked.us>

On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
> Hi,
> 
> I'm trying to get started with the ESL using PHP. I compiled the ESL, then
> phpmod according to the wiki instructions, but then when I try the examples
> in the libs/esl/php dir, they fail saying:
> 
> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
> 
> Checking ESL.php on line 132, I see there are several different declarations
> for the function __construct() with different parameters each, which makes
> sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
> higher than that or something? What could be the problem?
>

Someone in the IRC channel mentioned this too. I looked at it briefly
and it looks like the latest 'swigall' screwed it up. The original
reporter said he'd file a jira, but you may want to check yourself and
if not make one yourself. In the meantime, the previous version of the
file was reported to work if you really need it.

Andrew


From neffs1 at gmail.com  Fri Aug  7 07:02:13 2009
From: neffs1 at gmail.com (David Kreitschmann)
Date: Fri, 7 Aug 2009 16:02:13 +0200
Subject: [Freeswitch-users] Compact, fanless appliance?
In-Reply-To: <7d0bfd8c0907051621u1c3f553fh96a9df8557952e47@mail.gmail.com>
References: <7d0bfd8c0907041941s75e541dex982fae2195858ea4@mail.gmail.com>
	<C6757EE0.5891C%krice@suspicious.org>
	<7d0bfd8c0907051621u1c3f553fh96a9df8557952e47@mail.gmail.com>
Message-ID: <149B556F-65B6-40F7-802B-92B81447C24F@gmail.com>

I am thinking about using this combination as a router, should also be  
a nice platform for freeswitch

http://www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm
http://www.cartft.com/catalog/il/1058
http://www.cartft.com/catalog/il/1087

energy efficient netbook chipset (most other boards use the desktop  
version), PCI slot, small case, fanless.
you can put the system on an usb stick and just hide it behind the  
front panel. no need for reinstall on the device itself, just put in a  
another usb stick and you're good to go.

or you put in two disks in raid1 for redundancy if you need the disk  
space.

if you don't need PCI you can use this enclosure
http://www.cartft.com/catalog/il/1081


David

Am 06.07.2009 um 01:21 schrieb Nandy Dagondon:

> ok. w/ my apologies. - nandy
>
>
> On Sun, Jul 5, 2009 at 10:49 AM, Ken Rice <krice at suspicious.org>  
> wrote:
> No need to bump these things as this is a mailing list and it annoys  
> quite a few people when you do that
>
>
> From: Nandy Dagondon <gcd at i.ph>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 5 Jul 2009 10:41:18 +0800
>
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Compact, fanless appliance?
>
> just bumping this topic.
> -nandy
>
> On Fri, May 8, 2009 at 12:44 AM, Fred-145 <codecomplete at free.fr>  
> wrote:
>
>
> Antonio Gallo wrote:
> > Alix cases are like 6/9 ? from their shop site. I think its easy  
> to find
> > someone who work with aluminium that can make for you custom boxes  
> for
> > like like 6/20 ? at pcs
>
> Unfortunately, none of the PCEngines cases (www.pcengines.ch/order1.php?c=2 
>  <http://www.pcengines.ch/order1.php?c=2> )
>
> allow for a PCI slot, either on top of the mobo, or away from it :-/
>
> I'll see if I can get those from Soekris (http://soekris.eu/shop/cases_en/ 
> )
> allow this, and if I can get a good price for a case + PSU.
>
> Thank you.
> --
> View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fraunhofer.lists.freeswitch-001 at traced.net  Thu Aug  6 13:28:19 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Thu, 6 Aug 2009 22:28:19 +0200
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
Message-ID: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>

Hello List, Hello *,

First of all the usual excuses: sorry for the bad english and the long
email, no native speaker and i really tried to make it shorter, but i
guess this would result in even more "check back"s than it already
does :)

we're currently running in a weird "lockup"-scenario in our loadtests.

Our setup is the following:

three freeswitch servers, let's call them A(-leg), M(aster), and
B(-leg) with the goal in mind to initiate calls on M which calls A,
play some file, bridge to B, limit call length and play (different)
prompts to A and B if they exceed that limit.

(Note that A and B work fine, regardless of the amount of load we put on them)
A and B are silly dialplan logic, accepting calls on a certain
extension after a random delay and playing moh. Before calling
playback to a localstream they call a lua script which schedules
hangup somewhere in future (which works flawlessly)

Calls are initiated on M using some hacked up loadgen-script issuing
http requests like
? originate [sofiaSyntaxToExtensionOn_A] 6000
. The 6000 extension on M has the following (xml) dialplan which
essentially does the following:
------
answer()
...playback file...
...set some callerid stuff
set bypass_media
bridge to extension 6009 on B
------
we use "execute_on_answer" on the b-leg to run a script which limits
the length of the call (doesn't matter if it's done via "export
nolocal" or "inlined" into the data part of the bridge application
"{execute_on_answer=lua ...}")

<action application="export" data="nolocal:execute_on_answer=lua
lua/schedule-hangup.lua ${uuid}" />

the lua script "schedula-hangup.lua" does essentially the following:

------
api = freeswitch.API();
local res = api:execute("sched_api", "+10 none lua
lua/c2c-hangup-timeout.lua " .. argv[1]);
------

the 10 seconds are just to speed up the time until it gets stuck.

this is where things start to go wrong. if I comment out the call to
the "schedule-hangup" script, everything works fine, even if it's
under heavy load.

c2c-hangup-timeout.lua does the following:
------------------
local sess = argv[1];
if(sess)
then
? ?freeswitch.consoleLog("INFO", "c2c-hangup-timeout.lua for uuid "
.. sess .. "\n");

? ?api = freeswitch.API();
? ?local stillValid = api:execute("uuid_getvar", sess .. "
Dummy-DoesChannelExists");
? ?if(stillValid:sub(1,4) == "-ERR")
? ?then
? ? ? ?log("session uuid " .. sess .. " disappeared (nothing bad)");
? ?else
? ? ? ?-- this is important!!! Otherwise the aleg get's just hung up!
? ? ? ?api:execute("uuid_media", sess);
? ? ? ?api:execute("uuid_transfer", sess .. " -both timeout");
? ?end
else -- /if(sess)
? ?log("called with nil session?");
end -- /if(sess)

------------------

i guess this needs some explanation:
we get the uuid of the channel as argument in argv[1]. We don't use
? local session = freeswitch.Session(uuid);
since if the channel referenced by "uuid" does not exist any longer,
freeswitch (or the lua bindings) try to interpret the uuid as an
"originate string" and can't figure out how to call that. So we use a
dummy api call to get some channel variable. If the channel does not
exist any longer (A or B already hung up), we get an error message
starting with "-ERR", otherwise the channel still exists (we get
"_unset_" as the value, if it's not set) and we continue by getting
freeswitch back in the media path (uuid_media) and then transferring
both legs to an extension called "timeout" which plays some prompt and
finally calls hangup().

If we don't do the uuid_media call, one of the legs gets hung up when
we transfer them to the extension. This looks like the following on
the console after issuing "uuid_transfer [uuid] -both timeout"
(extensions are not the same as in our loadgen example above)


--------------
2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1334 Hangup (*)
sofia/internal/1000 [CS_HIBERNATE] [BLIND_TRANSFER]
2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer
sofia/internal/1004 at 192.168.179.177:5060 to XML[timeout at default]
2009-07-23 19:57:19.865703 [INFO] mod_dialplan_xml.c:310 Processing
BFR1004->timeout in context default
API CALL [uuid_transfer(73812082-77b1-11de-b9f8-a10bb0eb9f69 -both
timeout)] output:
+OK

2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer (**)
sofia/internal/1000 to XML[timeout at default]
2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1084 Session
60 (sofia/internal/1000) Ended
2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1086 Close
Channel sofia/internal/1000 [CS_DESTROY]
-----------

note that it first does Hangup (denoted by *, no that's not an
asterisk :) on extension 1000 and then tries to Transfer (**) the hung
up channel to the dial plan. this could be the same as in an earlier
post to the list "SIP re-invite / bypass_media // Phillip Jones //
Wed, 01 Jul 2009 13:30:53 -0700)"

This is why we do not directly call sched_transfer() but call a script
in between to do the uuid_media() call. I couldn't figure out how to
call that directly from the xml dialplan and/or how to check if the
channel still exists.

so... after using uuid_media(), both legs are transferred without an
(intermediate|bogus) hangup() call.

This only works fine if we've few concurrent calls. There is no magic
borderline where it starts to refuse work.

Some of the Symptoms are: traffic decreased to zero as no new channels
are successfully brought up, some of the signaling traffic is not
ACKed or OKed, scheduled jobs are not run.

if i read the output of "show channels" correctly, they're all stuck
in different applications like hangup(), some are calling lua but most
of them are in signaling_bridge(). Freeswitch is still responding on
the console and there's almost no load on the machine (no busy polling
or some other kind of running amok).

if i kill one of them using uuid_kill() or kill all of them
using"fsctl hupall" i get "Task was executed late by 866 seconds 12379
sched_api_function (none)" messages and the usual cleanup takes place.
As a quick hack i tried to schedule a uuid_kill() call 20 seconds
after the scheduling call to the lua script but that job is not
executed either.

So what am I doing wrong? Is it some deadlock where uuid_media() and
uuid_transfer() ?are waiting for the other to finish?
Or some other silly simple thing i missed?

Thx in advance

?Benedikt.


From email.list.subscriber at gmail.com  Thu Aug  6 15:03:56 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Thu, 6 Aug 2009 18:03:56 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
Message-ID: <4a7b530a.29578c0a.53a8.0450@mx.google.com>

Hello,

Does anyone have, or know where to get, a pre-compiled copy of
FreeSwitch for Solaris 10/x86?

# uname -a
SunOS hrndvsoi-zm01 5.10 Generic_137112-07 i86pc i386 i86pc


I'm stuck trying to run make/gmake/\/opt/gnu/bin/make:

<snip>
"make" results in:
make: Fatal error: Command failed for target `all-recursive'
Current working directory /home/vmorales/freeswitch
*** Error code 1
make: Fatal error: Command failed for target `all'
</snip>


"gmake" & "/opt/gnu/bin/make" result in:

<snip>
creating libfreeswitch.la
(cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la
libfreeswitch.la)
gcc -I/home/vmorales/freeswitch/src/include
-I/home/vmorales/freeswitch/libs/libteletone/src -fPIC -Werror -g
-ggdb -DPATH_MAX=2048 -g -m32 -I/usr/sfw/include -Wall -std=c99
-pedantic -m32 -o .libs/freeswitch freeswitch-switch.o  -L/usr/sfw/lib
-lm ./.libs/libfreeswitch.so
/home/vmorales/freeswitch/libs/apr/.libs/libapr-1.a
-L/home/vmorales/freeswitch/libs/srtp /usr/sfw/lib/libstdc++.so
libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt -ldl -lnsl -lpthread
libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lcurses -lsocket
-R/home/vmorales/freeswitch-build/lib -R/usr/sfw/lib
Undefined                       first referenced
 symbol                             in file
XML_Parse                           ./.libs/libfreeswitch.so
XML_ParserCreate                    ./.libs/libfreeswitch.so
XML_ErrorString                     ./.libs/libfreeswitch.so
herror                              ./.libs/libfreeswitch.so
XML_SetUserData                     ./.libs/libfreeswitch.so
XML_ParserFree                      ./.libs/libfreeswitch.so
XML_GetErrorCode                    ./.libs/libfreeswitch.so
XML_SetCharacterDataHandler         ./.libs/libfreeswitch.so
XML_SetElementHandler               ./.libs/libfreeswitch.so
ld: fatal: Symbol referencing errors. No output written to
.libs/freeswitch
collect2: ld returned 1 exit status
gmake[2]: *** [freeswitch] Error 1
gmake[1]: *** [all-recursive] Error 1
gmake: *** [all] Error 2
</snip>


Thanks in advance for any information provided.

Vladimir



From alan at chandlerfamily.org.uk  Fri Aug  7 01:18:35 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Fri, 07 Aug 2009 09:18:35 +0100
Subject: [Freeswitch-users] New to Freeswitch - some help needed
Message-ID: <4A7BE35B.8010709@chandlerfamily.org.uk>

I apologize, as my first post to this list, that I ask a detailed set of 
questions, but I have spend some time looking at all the docs and can't 
get what I need to do completely sorted in my head.  I am definitely one 
who likes to UNDERSTAND what is happening rather than follow blank 
recipies, so please bear with me as I try understand all the details. I 
do understand about networking, NAT etc - but I am new to SIP/RTP and in 
particular what I think is a double NAT problem


Firstly - what am I trying to achieve:

I am in the UK and have a small home network behind a D-Link DIR-100 
Router/NAT/Firewall one of those machines, running Debian Lenny, acts as 
my main server for everything (and in an earlier incarnation was the 
firewall/router/nat box too - I only say this is because I had all this 
working using Asterisk a year or so ago, but with this important 
difference in configuration).  Many of the ports on the firewall are 
port forwarded to this machine. I have set Freeswitch up on this server 
to act as a small voip pbx for the home - but MORE IMPORTANTLY - to 
enable my daughter from her house to talk to us.  At my house locally I 
have a Linksys PAP2T two phone SIP box - and that is working with 
Freeswitch's default configuration (I set up to be 1000 and 1001 and 
used all the facilities).  I will later add a Linksys SPA 3102 - 
although I DO NOT intend to use its facility to bridge to the normal 
phone network.

My daughter, living in another house, also has a Nat box (unknown - its 
part of her ADSL modem/router/wireless access point) and also has a 
PAP2T which she will connect to the her network.  This will be her phone.

There is a family relation living in Australia who will load up a 
whatever softphone that we tell him to use.  I expect, but don't know, 
that he will behind a NAT box too.

Later, I have some friends in the USA that I might wish to add it too - 
especially so that we can hold some teleconferences.  They will have a 
mixture of Windows and MACs, and I will need to recommend softphone 
clients for them.

I want to set this up as a small private voice network, so anyone can 
ring anyone else.  I will add fancy facilities such as conferencing and 
voicemail later - I just want to get the basics working first.

Secondly

I installed a stun client on my home machine and ran it against 
stun.freeswitch.org.

It reported:-

Primary: Independent Mapping, Independent Filter, preserves ports, no 
hairpin

But I have no idea what this means - I can't find any clear statement 
via googling for it - how this set of answers maps to the different 
types of NAT that might be required to get this to all work.  CAN 
SOMEONE ENLIGHTEN me please.

Thirdly

I have set up a sip profile called "double nat" from the recipe in the 
wiki.  This defines the SIP port to be 5090.

However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters 
house will initiate a connection to my server.  Presumably, I have to 
port forward 5090 from the nat box to my server.  IS THAT CORRECT?

I also assume I will have to tell her to use STUN (I believe this is an 
option on the PAP2T)

Fourthly

If I understand SIP correctly, it just initiates the session and the two 
end points then communicate directly via RTP.  What I don't understand 
is how does a session transition from SIP to RTP via the connection set 
up in the the first phase (in terms of passing through the NAT boxes). 
In particular WHICH OF THE TEST RESULTS from my stun client indicate it 
will do the right thing.  (I am going to take a laptop to my daughters 
house with a stun client in to test her network this weekend).

Could someone explain please.

  Fifthly

Is there a recommended SIP softphone with all the right facilities (STUN 
support?)  that works on MAC and WINDOWS (I only use linux myself).

Apologies for the length of this.  I am eager to get the answers so I 
can use an opportunity this weekend to get it working.


-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From alan at chandlerfamily.org.uk  Fri Aug  7 05:19:51 2009
From: alan at chandlerfamily.org.uk (alan at chandlerfamily.org.uk)
Date: Fri, 7 Aug 2009 13:19:51 +0100 (BST)
Subject: [Freeswitch-users] Auto Nat
Message-ID: <9f0cf614f2b2ffb9f7a2edae3bf3d2ff.squirrel@webmail.chandlerfamily.org.uk>

I sent my first e-mail to the list this morning (about 4 hours ago) but it
does not seem to have arrived back, even though I have received other,
later posts.

I have another question related to the first (about how to set everything
up in a double nat environment) - so if I see this and not the other, I
will send the first again.

I am currently running the stable version 1.0.3 of freeswitch.  The wiki
page says that auto-nat is introduced at r 13612.  Is this before or after
that revision? (I don't want to have to download and rebuild the entire
thing if I don't have to).






From alan at chandlerfamily.org.uk  Fri Aug  7 06:17:03 2009
From: alan at chandlerfamily.org.uk (alan at chandlerfamily.org.uk)
Date: Fri, 7 Aug 2009 14:17:03 +0100 (BST)
Subject: [Freeswitch-users] 1.0.4 builds 1.0.3_1 debs
Message-ID: <631613604d8c9862f1cf43f80cb99f94.squirrel@webmail.chandlerfamily.org.uk>

I just downloaded 1.0.4 and build debian packages with it, and it
delivered .deb files names as 1.0.3-1




From demuel at thephinix.org  Fri Aug  7 21:04:44 2009
From: demuel at thephinix.org (demuel at thephinix.org)
Date: Sat, 8 Aug 2009 05:04:44 +0100 (BST)
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <4A7BE35B.8010709@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
Message-ID: <1bfed71a616412ee86da3a7f18b990a0.squirrel@www.thephinix.org>

What a long detailed list of todos. You certainly can't find that kind of
answers in here.

> I apologize, as my first post to this list, that I ask a detailed set of
> questions, but I have spend some time looking at all the docs and can't
> get what I need to do completely sorted in my head.  I am definitely one
> who likes to UNDERSTAND what is happening rather than follow blank
> recipies, so please bear with me as I try understand all the details. I
> do understand about networking, NAT etc - but I am new to SIP/RTP and in
> particular what I think is a double NAT problem
>
>
> Firstly - what am I trying to achieve:
>
> I am in the UK and have a small home network behind a D-Link DIR-100
> Router/NAT/Firewall one of those machines, running Debian Lenny, acts as
> my main server for everything (and in an earlier incarnation was the
> firewall/router/nat box too - I only say this is because I had all this
> working using Asterisk a year or so ago, but with this important
> difference in configuration).  Many of the ports on the firewall are
> port forwarded to this machine. I have set Freeswitch up on this server
> to act as a small voip pbx for the home - but MORE IMPORTANTLY - to
> enable my daughter from her house to talk to us.  At my house locally I
> have a Linksys PAP2T two phone SIP box - and that is working with
> Freeswitch's default configuration (I set up to be 1000 and 1001 and
> used all the facilities).  I will later add a Linksys SPA 3102 -
> although I DO NOT intend to use its facility to bridge to the normal
> phone network.
>
> My daughter, living in another house, also has a Nat box (unknown - its
> part of her ADSL modem/router/wireless access point) and also has a
> PAP2T which she will connect to the her network.  This will be her phone.
>
> There is a family relation living in Australia who will load up a
> whatever softphone that we tell him to use.  I expect, but don't know,
> that he will behind a NAT box too.
>
> Later, I have some friends in the USA that I might wish to add it too -
> especially so that we can hold some teleconferences.  They will have a
> mixture of Windows and MACs, and I will need to recommend softphone
> clients for them.
>
> I want to set this up as a small private voice network, so anyone can
> ring anyone else.  I will add fancy facilities such as conferencing and
> voicemail later - I just want to get the basics working first.
>
> Secondly
>
> I installed a stun client on my home machine and ran it against
> stun.freeswitch.org.
>
> It reported:-
>
> Primary: Independent Mapping, Independent Filter, preserves ports, no
> hairpin
>
> But I have no idea what this means - I can't find any clear statement
> via googling for it - how this set of answers maps to the different
> types of NAT that might be required to get this to all work.  CAN
> SOMEONE ENLIGHTEN me please.
>
> Thirdly
>
> I have set up a sip profile called "double nat" from the recipe in the
> wiki.  This defines the SIP port to be 5090.
>
> However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters
> house will initiate a connection to my server.  Presumably, I have to
> port forward 5090 from the nat box to my server.  IS THAT CORRECT?
>
> I also assume I will have to tell her to use STUN (I believe this is an
> option on the PAP2T)
>
> Fourthly
>
> If I understand SIP correctly, it just initiates the session and the two
> end points then communicate directly via RTP.  What I don't understand
> is how does a session transition from SIP to RTP via the connection set
> up in the the first phase (in terms of passing through the NAT boxes).
> In particular WHICH OF THE TEST RESULTS from my stun client indicate it
> will do the right thing.  (I am going to take a laptop to my daughters
> house with a stun client in to test her network this weekend).
>
> Could someone explain please.
>
>   Fifthly
>
> Is there a recommended SIP softphone with all the right facilities (STUN
> support?)  that works on MAC and WINDOWS (I only use linux myself).
>
> Apologies for the length of this.  I am eager to get the answers so I
> can use an opportunity this weekend to get it working.
>
>
> --
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From jason at jasonjgw.net  Fri Aug  7 21:23:42 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 8 Aug 2009 14:23:42 +1000
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <4A7BE35B.8010709@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
Message-ID: <20090808042342.GA3558@jdc.jasonjgw.net>

Alan Chandler <alan at chandlerfamily.org.uk> wrote:
 
> I want to set this up as a small private voice network, so anyone can 
> ring anyone else.  I will add fancy facilities such as conferencing and 
> voicemail later - I just want to get the basics working first.

I have a similar arrangement operating here which involves friends and
colleagues in the U.S., as well as a local VoIP provider that gives me access
to the PSTN.

To eliminate NAT issues, we are using IPv6: each of us has an IPv6 over IPv4
tunnel configured to provide access to the IPv6 Internet. NAT and all the
problems associated with it go away.

Another option, although I don't know how well real-time communication works
in this setting, would be to create a VPN using, for example, OpenVPN so that
the clients and server all appear to be on the same lan.

Alternatively, you could play with port forwarding and FreeSWITCH settings in
an attempt to work around the nat issues - good luck!

I can't answer any questions about MacOS or Windows softphones - there are no
MacOS or Windows machines in my life.



From gcd at i.ph  Fri Aug  7 21:33:37 2009
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 8 Aug 2009 12:33:37 +0800
Subject: [Freeswitch-users] Auto Nat
In-Reply-To: <9f0cf614f2b2ffb9f7a2edae3bf3d2ff.squirrel@webmail.chandlerfamily.org.uk>
References: <9f0cf614f2b2ffb9f7a2edae3bf3d2ff.squirrel@webmail.chandlerfamily.org.uk>
Message-ID: <7d0bfd8c0908072133p7b7f68aancfabdb78d5fe7954@mail.gmail.com>

r 13612 is after 1.0.3. you better get 1.0.4 recently released.

-nandy

On Fri, Aug 7, 2009 at 8:19 PM, <alan at chandlerfamily.org.uk> wrote:

> I sent my first e-mail to the list this morning (about 4 hours ago) but it
> does not seem to have arrived back, even though I have received other,
> later posts.
>
> I have another question related to the first (about how to set everything
> up in a double nat environment) - so if I see this and not the other, I
> will send the first again.
>
> I am currently running the stable version 1.0.3 of freeswitch.  The wiki
> page says that auto-nat is introduced at r 13612.  Is this before or after
> that revision? (I don't want to have to download and rebuild the entire
> thing if I don't have to).
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri Aug  7 21:36:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 00:36:58 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
Message-ID: <84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>

This is not currently a supported platform, it only builds on 64 bit  
right now I think on solaris.

Mike

On Aug 6, 2009, at 6:03 PM, vmorales wrote:

> Hello,
>
> Does anyone have, or know where to get, a pre-compiled copy of
> FreeSwitch for Solaris 10/x86?



From mcampbellsmith at gmail.com  Fri Aug  7 21:39:14 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 8 Aug 2009 14:39:14 +1000
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <4A7BE35B.8010709@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
Message-ID: <33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>

Hi Alan,

I hope you find your answers here as these are the sort of things that
are hard to find on the wiki, which is somewhat outdated in areas.  If
you do find your answers, please post them back here for everyone
else.

I am new to FS also, so my comments below may